Académique Documents
Professionnel Documents
Culture Documents
1.About BSNL
2.About Alcatel
3.Description Of E-10B
4.Pulse Code Modulation
5.Conclusion
ABOUT BSNL
1
Bharat Sanchar Nigam Ltd. formed in October, 2000, is World's 7th largest
Telecommunications Company providing comprehensive range of telecom services
in India: Wireline, CDMA mobile, GSM Mobile, Internet, Broadband, Carrier service,
MPLS-VPN, VSAT, VoIP services, IN Services etc. Presently it is one of the largest
& leading public sector unit in India.
BSNL has installed Quality Telecom Network in the country and now focusing
on improving it, expanding the network, introducing new telecom services with ICT
applications in villages and wining customer's confidence. Today, it has about 46
million line basic telephone capacity, 8 million WLL capacity, 52 Million GSM
Capacity, more than 38302 fixed exchanges, 46565 BTS, 3895 Node B ( 3G
BTS), 287 Satellite Stations, 614755 Rkm of OFC Cable, 50430 Rkm of
Microwave Network connecting 602 Districts, 7330 cities/towns and 5.6 Lakhs
villages.
BSNL is the only service provider, making focused efforts and planned
initiatives to bridge the Rural-Urban Digital Divide ICT sector. In fact there is no
telecom operator in the country to beat its reach with its wide network giving services
in every nook & corner of country and operates across India except Delhi & Mumbai.
Whether it is inaccessible areas of Siachen glacier and North-eastern region of the
country. BSNL serves its customers with its wide bouquet of telecom services.
BSNL is numero uno operator of India in all services in its license area. The
company offers vide ranging & most transparent tariff schemes designed to suite
every customer.
BSNL has more than 2.5 million WLL subscribers and 2.5 million Internet
Customers who access Internet through various modes viz. Dial-up, Leased Line,
DIAS, Account Less Internet(CLI). BSNL has been adjudged as the NUMBER ONE
ISP in the country.
BSNL SERVICES
BSNL provides almost every telecom service in India. Following are the main
telecom services provided by BSNL:
• Intelligent Network (IN): BSNL provides IN services like televoting, toll free
calling, premium calling etc.
• IPTV:BSNL also offers the 'Internet Protocol Television' facility which enables
us to watch television through internet.
• FTTH:Fibre To The Home facility that offers a higher bandwidth for data
transfer.This idea was proposed on post-December 2009.
3
• Wi-MAX: In addition to wireline broadband services, BSNL is also in the
process of rolling out its Wi-MAX network in rural areas to take an initial lead
and provide wireless broadband services in all rural blocks in the country
during 2010-11. The Urban Wi-Max is also being deployed in Kerala & Punjab
Circles and shall cover all the mojor cities in these circles.
• Wi-Max services are also being provided through a Franchisee agent with M/s
SOMA in three states of Gujrat, AP and Maharashtra
ABOUT ALCATEL
History
Alcatel has a long history begininig in 1898 with the founding of Compagine
Generale d'Electricite(CGE). The original home of the company was the Alsace
region and it still maintains R&D operations in the Strasbourg area. The current
4
name of "Alcatel", comes from the acquisition in 1968 of Societe Alsacienne de
constructions atomiques,de Telecommunications et d' Electronique
Early in 2006, Alcatel setup a new joint venture with TCL of china forming a
new mobile business, TCL and Alcatel Mobile Phones Limited(TAMP).
Alcatel-Lucent
Alcatel-lucent is one of the world's biggest industryplayers in
telecommunications that provides hardware,software and services to service
providers and enterprise all over the globe. The company is incoprated in france and
has it headquaters at rue de la Boetie in Paris. The company does business in more
than 130 countries,with almost equal sales distribution coming from both its
European and North American regions, and an additional third of its channel located
elsewhere ni the world. Alcatel-Lucent was formed after Alcatel merged with Lucent
technologies on december 1,2006.
Areas of business
Alcatel was mostly well known for its DSL multiplexers, used for high speeed
internet access over ADSL and VDSL, wheras Lucent was well- known for its class 4
nad 5 voice switches (central office) and its optical products, Alcatel had over 40% of
the world DSLAM market in 2007, with more than 143 million lines shipped and has
been evolving this from an ATM-backhauled device to an IP-backhauld device. It has
a partnership with Microsoft as of 2004 to provide IPTV services via its TPSDA(Triple
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Play Servives Delivery Architecture) over DSL and using its 7*50 VPLS/MPLS
routers and switches to service provides such as AT&T in the united states. It also
leading provider of optical transmission equipment, especially for submarine
communication cable. Genesys, a U.S. subsidary, is a leading provider of a call
centre software which operates both with Alcatel-Lucent equipment and 3rd-party
equipment. The company is generally organised into carrier, enterprise and services
business groups.
7
transmission system. At present, E-10B electronic digital switching developed by
CIT Alcatel of France is being introduced into our network on a very large scale.
The system has several versions. The version supplied to India is the 384
PCM versions which can handle a maximum traffic of 4000 Erlangs.
The number of Busy Hour Call Attemps (BHCA) that the system can handle is
found 1, 90,000
(c) TAX
When used as a tax, the system provides for the termination of long
distance circuits. Digital TAX of the E-10B type have already been
commissioned at various stations. The maximum capacity of an E-10B
Digital TAX is limited to 1100 (O/G and I/C in the 384 PCM versions.
8
It is possible to combine the functions of local and transit or TAX
exchanges in an E-10B system.
9
Fig.1 CONCEPT OF RLU
The remote subscribers get all the facilities available to the local (main
E-10B subscribers. Cabling costs are reduced drastically since the
subscribers are now connected by short cable pairs straight to the RLUs
instead of to the distant main exchange. However, the URAD (CSED) has no
‘stand-alone’ capability, i.e. in case of failure of PCM between URAD and
parent exchange, the URAD is isolated, i.e. subscribers can neither make any
local or outgoing calls nor can receive incoming calls. Subscribers can
security call. This facility enables them to get access to predetermined
emergency services only viz. police, Fire and Ambulance by dialing special
codes.
The E-10B URADs are being progressively utilize to fulfill the telecom
needs of sparsely populated areas having a community of telecom interest
with main towns/urban centers.
10
6.0 Introduction of E-10B in the Analogue network (Fig.2)
The E-10B system can be introduced into an existing analog network with
ease. The analogue electromechanical systems e.g. strowger, crossbar local and
TAX system and analog electronic systems like PRX-A, ND-10, FETEX-100L need
interface equipments before being connected to the E-10B system. Two interfaces
are digital signaling acceptable by E-10B and the other to carry out analog to digital
(A to D) conversion (and vice versa). The signaling conversion is carried out by a
group of signaling converters known as GAS (Signaling Adapter Group). The A/D
and D/A conversion is performed by the PCM multiplexing equipment called TNE or
Digital Terminal Equipment. The equipments are situated ideally at the analogue
exchange ends. The output of TNE stream. The TNE/GAS equipments are situated
in the same rack. Each fully equipped 180 junctions.
11
7.0. Basic principles and architecture of E-10B system:
The system switches signaling digital form. Analog signals are converted into
Time-division multiplexed digital signals prior to switching.
12
The various call handling and call-processing functions status, receptions and
storage of digitas, analysis and routing, metering etc. are distributed over various
functional units. Dedicated micro processors INTEL 8085 and dedicated mini
computer ELS-4B handle these functions.
13
7.2. E-10B EXCHANGE ARCHITECTURE (Fig.5)
Appendix-I
CAPACITY
SYSTEM
15
SUBSCRIBER LINE:
ENVIRONMENTAL CONDITIONS:
• Exchange:
Ambient temperature of air drawn into racks: 18 to 20 deg. Celsius
Relative humidity: 30 to 70%
• Satellite exchange:
Ambient temperature: 5 to 35 deg. Celcius
Relative humidity: 20 to 80%
• OMC:
Air- conditioned environment
Temperature 15 to 18 deg. C (optimum 22 ±2 deg.C)
MECHANICAL DATA
FLOOR AREA
Rack dimensions
• Height: 2.00 m
• Width: 0.75
• Depth:0.50m
Distributed floor loading:
less than 500 kg/ sq. meter
POWER SUPPLY:
16
CONNECTION UNITS
1.0 The connection units of the E10B system as mentioned earlier provide the
required interface between subscribes/trunks and the E10B Central Units.
There are four types of connection units. These are:
To switching n/w
CSEL(URAL)
The CSE can also be located remotely from the exchange to economies on
outdoor plant investment, and then it is called a ‘Distant Subscribers connection Unit’
and is designated as URAD or CSED. The CSE is actually is Remote Line Unit
(RLU) and is connected to the parent (Main) E10B exchange by PCM links. (Fig. 2)
CSED URM
(URAD)
channels.
vii. Routine Subscriber line tests under the control or Operation and
Maintenance Center (OMC)
2.2 Subsystems of CSE : The CSE can be devided into two subsystems-
1. Singalling equipment
2. Concentration network
3. Transmission equipment.
The Singalling equipment is the Subscriber line card called XEJ. The purpose
of this unit is
To switching n/w
PCM – MODULE ‘0’
From CSE
When distant URs are used, code converters provide also the line coding
function (Binary to HDB3 and vice versa.) Each of the PCM modules are
connected to the switching network (CX) by 2 Mbit/sec PCM links Called LR
links. These Lr links can carry speech, tones and frequencies in the digital
form (Fig. 4).
The control logic system manages the speech path subsystem and also
ensures exchange of message with the central units of the E-10B exchange.
19
The logic system is duplicated and is provided by Intel 8085 microprocessors,
along with their memories. One of the logic system (Intel 8085) is on-line and
the other in hot-stand by mode. Either of the two logic units can be the active
logic at a given point of time. Only one logic has access to the speech path at
any point of time, while the other logic carries out tests and updates the
memories.
These PCM Systems terminate at the URM (Multiplex connection unit) of the parent
exchange. Regenerative Repeaters (RR) are provided along the PCM route. The
digital line terminal (TNL) provided at each end fulfills the functions specific to the
transmission medium (e.g. power feeding of repeaters, fault localization etc).
2.4 The first 2 PCM links emanating from the CSED are designated as PCM-o
and PCM-1 and are called the ‘Active PCMs and the other two PCMs
designated as PCM-2 and PCM-3 are called ‘passive PCMs’. Active PCMs
carry speech and signaling while passive PCMs carry, speech only. Signalling
for all the four all the four the four PCMs is sent on two active PCMs only.
2.5 The CSED also offers the possibility of setting ‘local security calls’ In the event
of a breakdown of the announcement informing him of the 2-digit number to
be dialed to obtain an emergency service (fire, police, ambulance etc.)
1. PCM synchronization
2. Signalling byte injection and extraction (TS-16)
3. Speech-path ‘mixing’
4. Fault localization
3.1 The multiplex Connection Unit is the PCM connection interface between E-
10B exchange and distant subs terminate upto thirty-two, 30-Channel PCMs
i.e., 960 junctions or circuits can be terminated.
Both module types may coexist within the same URM in any required
proportion.
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The URM logic consists of one LOGUR system duplicate (LOGUR-0 and LOGUR-1)
for security reasons. Each LOGUR consist of a Main (Master) Processor and two
Auxiliary (Slave) Processors a & B which are standard intel 8085 microprocessors.
Each LOGUR can function as Pilot or as Hot-standby unit keeps its memories
updated in real time to conform to control (pilot) logic. Logic switch over can take
place under the following contingencies.
3.2 In case the URM is connected to PCM links incoming from electromechanical
exchanges, analogue to digital conversion is necessary and in some cases
signaling conversion is also required. TNE is used for A/D conversion and
vice versa, while GAS converts DC loop signaling into E&M (digital) signaling
and vice versa.
receivers and
21
3.3.2 An E10-8 exchanges is equipped with a minimum of two and maximum of 16
ETA’s.
All the exchange units in the E-10B system are interconnected by dedicated
links carrying message for call processing.
LT links are connected to the URA and URM. These carry loop status of
subscribers (viz off-hook) and the status of circuits (ie idle, selzure etc). To
allow for all possible exchange configurations, 192 LT links are provided for
URA’s and URMs.
There are three types of links between the connection units and the switching
network. These are LRE, LRS and LVS links.
LRE and LRS links provide the path for speech/tone samples. There are 4
LRE links for each CSEL and for each module of URM (MRS/MRM).
1.0 CX Environment
22
In E-10B System, the switching network CX is connected to the Connection
Units, To the Markers (in particular to the UGCXs ie the Switching Network
Control Units associated with the marker) and the monitoring unit OC.
- The CX (1) interconnects calling and called timeslots and enables transfer of
speech samples between them.
- It directs DTMF (Dual Tone Multi Frequency i.e. Push button) frequencies
from subscribers and R2 (2 out of 5 MF) frequencies from circuits towards
ETA., for decoding into digits/appropriate signals.
1.2.1 Switching in CX
A connection in the switching network for any call will fall in one of the four
categories.
Calling party’s timeslot on the LRE- Calling party’s timeslot on the appropriate
LRS, where LRE and LRS are the input and output PCM highways terminated
in the CX. Such Connection are possible via the CX since the and receive
paths (LRS) for the speech samples. It is obvious that two timeslot are
required per call.
23
Max no. of LRs = 384
= 368
A switching efficiency of 80% i.g. traffic of O.B. Erlang per Timeslot can be
assumed.
11040 X 80
= 5520 x 0.8
= 4416
2.0 Modularity of CX
Each LR can carry speech samples. Each swathe module consists of one
input time stage (CTE), one space stage (CS) and one output time stage
(CTS).
The entire RCX is thus constituted of 24 Input Time stages, 24 space stages
and 24 output Time stages. The various stages of each switch module and
those of there is full flexibility for interconnecting the switch module can be
connected to any timeslot of any LRS located in any switch module (Fig. 2)
Each time stage is provided with its own buffer and control memories. the
space stage is equipped with a multiplexer type switch matrix and associated
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control memories. The timing signals needed for the read/ write operations
are derived from the mean time base(BT).
Normally the even numbered switch modules (0,2,…………22) egt their tones,
freq, supplies etc from the first ETA and the odd number ones (1,3………..23)
are supplied with tones etc by the second ETA. However in the case of failure
of one ETA the second ETA provides thetone supply to all the switch modules
for distribution. Thus, the first ETA ( ETA-1) acts as a normal source for a
even switch modules and as stand by source for odd switch modules and
vice- versa.
5.0 CX cpmfogiratopm
6.1 the switching network control unit known as UGCX is located in the
marker rack. UGCXO is associated with MQ1 and UGCX 1 with MQ2. the
UGCX is connected to marker by an internal data bus and by dedicated
links to CX,MR.TX.
25
3. it receives circuit signaling commands from MR and then sends
them to CX for outer transmission to circuits via URM
Architechture of UGCX
The UGCX has its own high speed processor for executing above
procedure with a instruction cycle time of 2 microseconds. The instructions
are of 4B Bits but only 32 bits are used. The program memory is organized in
a 4k word ‘sarcler’ card. Besides, the processor has a work memory,
instruction decoders etc.
CONTROL UNITS
1.0 dedicated control units carry out various operations related to call
processing. Some of these operations are:
reception and storageof dialed digits, analysis and digits, control over a
transmission of various tones, path finding via the switching network,
metering the call etc. there are 6 control units. These are
1. Marker MQ
2. Multiregister MR
3. Translater TR
4. Charging unit TX
6. Monitoring unit OC
The call processing and control functions are distributed over the 6
control units. These 48 provided in control units.
Marker (MQ)
The marker acts as a message distributer and routes the message associated
with call from one unit to other. It also finds a path through switching.
A marker is seized with:
Connection units in case of a new call or ‘ on hook’ condition ( i.e release) of a
subscriber
26
Multiregisters (1) for free/ busy/ check of subs ( URA)/ cct/ ( URM)/ frequency
receiver (ETA). (2) for connections as well as release of connections in sw. network
Two markers are provided in a E10B system. These two operate in load
sharing mode. Each marker occupies a rack which also houses a switching network
control unit (UGCX). UGCX controls the operations of switching networkon the basis
of messages received from control units like marker multiregister,charging unit etc.
The multiregister is the heart of the system ad performs overall command and
control functions for call set up and call release.
The multiregister:
- accepts seizer requests and on/ off hook conditions from subscribers and
circuits detected by connecting units.
- stores dialed digits.
- orders connection between time slots of calling and called parties in the
switching networks.
- orders transmission of tones, frequencies and signaling towards subscribers
and circuits.
- orders the release of calls.
27
These two sections exchange messages via buffer memory accessible by the
ELS of the interchange units by the processor of register units. The buffer is called
the I/Q buffer (TES).
OPERATING PRINCIPLE
It is a multiregister, any one out of 254 registers is used to process data for
Setting up or release of a call data required for setting up the call is stored in a
register of 64 words capacity during its seizure( when calling party goes off hook)
and is updated in usual time during call set up phase ( ex calling subscriber dialling
digits ). These registers are processed cyclically on a time sharing basis. A register
processing system at least once in every tocll set up or release is processed a
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number of times by register processing system executing a number of circuit
signaling stored in a RSI module is also made use of while processing. During call
set up/ release, exchange and depending on the phase of call sends sutaible
message for a call set up/ release.
4.0 TRANSLATOR
Data relating to the subscribers i.e. category, class of service and other
characterstics required by multiregisters during call processing are supplied by
translator. Similar data relating to circuits also stored in a translator. These data are
organised in a data memory in a translator in the form of lines. Man machine
commands from the OMC depending upon the even by subscribers ( eg. Data
relating to additional services) there are two translators which work on load sharing
basis.
This is a auxillary memory which stores all the data needed to set up and
release calls. The data memory (MATR) consists of maximum three memory
modules. Depending an amount of data to be stored , the translator will be equipped
one, two or three modules. Each memory module can store 256 k words ( 1 word =
16 bits) of data. The data in the memory is organized in the form of files In a typical
E- 10B exchange, the data memory consists of a maximum of 523 Files.
The translator is called by multiregister during call set up and its release.there
are two phases of call set up when translator is called up by multiregisters i.e.
preselectionand selection.
4.2.1 Preselection
4.2.2 Selection
- selection address( eg. UR no. and eqpt of the URA of the called sub.)
- charging data.
- characterstics of the circuit group to be used ( in the case of O/G
calls).
Any selection ( or translation) procedure in the translator may be represented as
follows.
The charging unit is organised in the same way as multiregister. There are three
Functional subsystems:
- Interchange unit
- Register unit
- I/O interface
5.1.1 The unit is organized around an ELS 4B processor. There are interchange
modules for various links. It includes a clock module (HORE) which provides a day,
hour minutes and seconds needed by ELS 4B processor to manage a charge rate
table as appropriate to the type of day and time. This information is also useful in
detailed data; it as also provided to translator for changing the route of time-
dependent special services( eg. Calls to ‘198’ can be routed to local Xge during
30
working hours and to a centralized location during slack hours). The interchange
unitalso contains a Auxillary charging memory (MATX).
• When calling party lifts the hand set, a charging unit is seized by mul
tiregister. This charging unit also calculate the charge and update
charge account in the other one at the end of the call.
- Data save
- Regeneration
REGENERATION
Each exchange has its own time base, which is not duplicated. The time base
gen erates the basic signals ( 2w, 20, h, ti). The timing signals are sent to
other exchange units where secondry. Timing signals are required for
synchronizing and co-ordinating various operations of E-10B exchange units.
32
For security reasons the signal generation system generating the basic timing
signals is triplicated 6.144 MHz oscillator is used for signal generation.
Provision exists in time base for fault detection & simulation.
The power distribution columns(MDE) are end of each suite. The MDE distributes
racks of its suite and provides voltage using fuses and circuit breadkers.
Functions of OMC:
• Subscriber management
• Trunk testing
• Unit positioning
• Fault tracing
• Fault clearing
33
Stages In A Local Call
There are three stages in the setting up of a local call. These are:-
1. Preselelection
2. Selection
1. Pre-Selection:
2. Selection:
Selection is the second main stage in the local call set upand
commences with subs dialing. This stage consists of five phases of
operation. These are:
a) Digit reception
c) Digit analysis/Translation
This is the third main stage in the local call set up and consists of the
following operations:
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d) Connection of calling party and called party time-slots.
e) Charging in TX.
Pulse-code modulation
Pulse-code modulation (PCM) is a digital representation of an analog signal where
the magnitude of the signal is sampled regularity at uniform intervals, then quantized
to a series of symbols in numeric (usually binary) code. PCM has been used in
digital telephone systems and 1980s-era electronic musical keyboards. It is also the
standard from for digital audio in computers and the compact disc “red book” format.
It is also standard in digital video, for example, using ITU-R BT.601. However,
straight PCM is not typically used for video in standard definition consumer
applications such as DVD or DVR because the bit rate required is far too high.
Modulation
In the diagram, a sine wave (red curve) is Sampled and quantized for PCM. The sine
wave is sampled at regular intervals, shown as ticks on the x-axis. For each sample,
one of the available values (ticks on the y-axis) is chosen by some algorithm (in this
case, the floor function is used). This produces a fully discrete representation of the
input signal (shaded area) that can be easily encoded as digital data for storage or
manipulation. For the sine wave example at right, we can verify that the quantized
values at the sampling moments are 7, 9, 11, 12, 13, 14, 15, 15, 15, 14, etc.
35
Encoding these values as binary numbers would result in the following set of nibbles:
0111, 1001, 1011, 1100, 1101, 1110, 1111, 1111, 1111, 1110, etc.
There are many ways to implement a real device that performs this task. In real
system, such a device is commonly implement on a single integrated circuit that
lacks only the clock necessary for sampling, and is generally referred to as an ADC
(Analog-to-digital converter). These devices will produce on their output a binary
representation of the input whenever they are triggered by a clock signal, which
would then be read by a processor of some sort.
Demodulation
To produce output from the sampled data, the procedure of modulation is applied in
applied in reverse. After each sampling period has passed, the next value is read
and the output of the system is shifted instantaneously (in an idealized system) to
the new value. As a result of these instantaneous transitions, the discrete signal will
have a significant amount of inherent high frequency energy, mostly harmonics, the
signal would be passed through analog filters that suppress artifacts outside the
expected frequency (see square wave). To smooth out the signal and remove these
undesirable harmonics, the signal would be passed through analog filters that
suppress artifacts outside the expected frequency range
(i.e. greater than, the maximum resolvable frequency). Some system use digital
filtering to remove the lowest and largest harmonics. In some system, no explicit
filtering is done at all; as it’s impossible for any system to reproduce a signal with
infinite bandwidth, inherent losses in the system compensate for the artifacts – or the
system simply does not require much precision. The sampling theorem suggests that
36
practical PCM devices, provided a sampling frequency that is sufficiently greater
than that input signal, can operate without introducing significant distortions within
their designed frequency bands.
The electronics involved in producing an accurate analog signal from the discrete
data are similar to those used for generating the digital signal. These devices are
DACs (digital-to-analog converters), and operate similarly to ADCs. They produce on
their output a voltage or current (depending on type) that represents the value
presented on their inputs. This output would then generally be filtered and amplified
for use.
Limitations
There are two sources of impairment in PCM system:
• Choosing a discrete value near the analog signal for each sample
(quantization error)
The quantization error swing between –q/2 to q/2 so mean = integration ( xf(X) dx )
the int from –q/2 to +q/2 which equal zero variance = int (x-mean)^2 f(x) dx the int
from –q/2 to +q/2 which equal to q^2/12.
Some forms of PCM combine signal processing with coding. Older version of these
system applied the processing in the analog domain as part of the A/D process,
newer implementation do so in the digital domain. These simple techniques have
been largely rendered obsolete by modern transform-based audio compression
techniques.
37
• Differential (or Delta) pulse-code modulation (DPCM) encodes the PCM
values as differences between the current and the predicted value. An
algorithm predicts the next sample based on the previous samples, and the
encoder stores only the difference between this prediction and the actual
value. If the prediction is reasonable, fewer bits can be used to represent
the same information. For audio, this type of encoding reduces the number
of bits required per sample by about 25% compared to PCM.
• Adaptive DPCM (ADPCM) is a variant of DPCM that varies the size of the
quantization step, to allow further reduction of the required bandwidth for a
given signal-to-noise ratio.
In telephony, a standard audio signal for a single phone call is encoded as 8000
analog samples per second, of * bits each, giving a 64 kbit/s digital signal known as
DS0. The default signal compression encoding open a DS0 is either u-law (mu-law)
PCM (North America and Japan) or a-law PCM (Europe and most of the rest of the
world). These are logarithmic compression system where a12 or 13 bit linear PCM
sample number is mapped into an * bit value. This system is described by
international standard G.711. An alternative proposal for a floating point
representation, with % bit mantissa and 3 bit radix, was abandoned.
Where circuit costs are high and loss of voice quality is acceptable, it sometimes
makes sense to compress the voice signal even further. An ADPCM algorithm is
used to map a series of 8 bit µ-law (or a-law) PCM samples into a series of 4 bit
ADPCM samples. In this way, the capacity of the line is doubled. The technique is
detailed in the G.726 standard.
Later it was found that even further compression was additional standards were
published. Some of these international standards describe systems and ideas which
are covered by privately owned patents and thus use of these standards requires
payments to the patent holders.
In other cases, the long term DC value of the modulated signal is important, as
building up a DC offset will tend to bias detector circuits out of their operating range.
In this case special measures are taken to keep a count of the cumulative DC offset,
and to modify the codes if necessary to make the DC offset always tend back to
zero.
Many of these codes are bipolar codes, where the pulses can be positive, negative
or absent. In the typical alternate mark inversion code, non-zero pulses alternate
between being positive and negative. These rules may be violated to generate
special symbols used for framing or other special purposes.
Theory
For convenience, we will discuss signals which vary with time. However, the same
results can be applied to signals varying in space or in any other dimension.
x[n]=x(nT),with n=0,1,2,,…
We can now ask: under what circumstances is it possible to reconstruct the original
signal completely and exactly (perfect reconstruction)?
39
A partial answer is provided by the Nyquist-Shannon sampling theorem, which
provides a sufficient (but not always necessary) condition under which perfect
reconstruction is possible. The sampling theorem guarantees that band limited
signals (i.e., signals which have a maximum frequency) can be reconstructed
perfectly from their sampled version, if
The sampling rate is more than twice the maximum frequency. Reconstruction in this
case can be achieved using the Whittaker- Shannon interpolation formula.
The frequency equal to one-half of the sampling rate is therefore a bound on the
highest frequency that can be unambiguously represented by the sampled signal.
This frequency (half the sampling rate) is called the Nyquist frequency of the
sampling system. Frequencies above the Nquist frequency fn can be observed in the
sampled signal, but their frequency is ambiguous. That is, a frequency component
with frequency f cannot be distinguished from other components with frequencies
NfN+f and NfN-f for nonzero integers N. This ambiguity is called aliasing. To handle
this problem as gracefully as possible, most analog signals are filtered with an anti-
aliasing filter (usually a low-pass filter with cutoff near the Nyquist frequency) before
conversion to the sampled discrete representation.
Sampling interval
Observation period
The observation period is the span of time during which a series of data samples are
collected at regular intervals. More broadly, it can refer to any specific period during
which a set of data points is gathered, regardless of whether or not the data is
periodic in nature. Thus a researcher might study the incidence of earthquakes and
tsunamis over a particular time period, such as a year or a century.
The observation period is simply the span of time during which the data is studied,
regardless of whether data so gathered represents a set of discrete events having
arbitrary timing within the interval, or whether the samples are explicitly bound to
specified sub-intervals.
Practical implication
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from the theoretically perfect reconstruction capabilities collectively referred as
distortion.
Aliasing: A precondition of the sampling theorem is that the signal be band limited.
However, in practice, no-limited signal can be band limited. Since signals of interest
are almost always time-limited (e.g., at most spanning the lifetime of the sampling
device in question), it follows that they are not band limited. However, by designing a
sampler with an appropriate guard band, it is possible to obtain output that is as
accurate as necessary.
Integration effect or aperture effect: This results from the fact that the sample is
obtained as a time average within a sampling region, rather than just being equal to
the signal value at the sampling instant. The integration effect is readily noticeable in
photography when the exposure is too long and creates a blur in the image. An ideal
camera would have an exposure time of zero. In a capacitor-based sample and hold
circuit, the integration effect is introduced because the capacitor cannot instantly
change voltage thus requiring the sample to have non-zero width.
Slew rate limit error, caused by an inability for an ADC output value to change
sufficiently rapidly.
Error due to other non-linear effects of the mapping of input voltage to converted
output value (in addition to the effects of quantization).
The conventional, practical digital-to analog converter (DAC) does not output a
sequence of dirac impulses (such that, if ideally low-pass filtered, result in the
original signal before sampling) but instead output a sequence of piecewise constant
values or rectangular pulses. This means that there is an inherent effect of the zero-
order hold on the effective frequency response of the DAC resulting in a mild roll-off
of gain at the higher frequencies (a 3.9224 db loss at the Nyquist frequency).This
zero- order hold effect is a consequence of the hold action of the DAC and is not due
to the sample and hold that might precede a conventional ADC as is often
misunderstood. The DAC can also suffer errors from jitter noise, slewing and non-
linear mapping of input value to output voltage.
Jitter, noise and quantization are often analyzed by modeling them as random errors
added to the sample values. Integration and zero-order hold effects can be analyzed
as a form of low-pass filtering. The non-linear ties of either ADC or DAC are
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analyzed by replacing the ideal linear function mapping with a proposed nonlinear
function.
Discrete signals (a common mathematical model) need not be quantized, which can
be a point of confusion. See ideal sampler.
Mathematical description
Q(χ)=g(└f(χ)┘)
Where
└.┘ is the floor function, yielding an integer result i= └f(χ) that is sometimes
referred to as the quantization index,
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In computer audio and most other applications, a method known as uniform
quantization is the most common. There are two common variations of uniform
quantization, called mid-rise and mid-tread uniform quantizes.
In this case the f(x) and g (i) operators are just multiplying scale factors (one
multiplier being the inverse of the other) along with an offset in g(i) function to place
the representation value in the middle of the input region for each quantization index.
The value 2 –(m-1) is often referred to as the quantization step size. Using this
quantization law and assuming that quantization noise is approximately uniformly
distributed over the quantization step size (an assumption typically accurate for
rapidly varying x or high M) and further assuming that the input signal x to be
quantized is approximately uniformly distributed over the entire interval from -1 to 1
the signal to noise ratio (SNR) of the quantization can be computed as
From this equation, it is often that the SNR is approximately 6 db per bit.
For mid-tread uniform quantization, the offset of 0.5 would be added within the floor
function instead of outside of it.
Sometimes, mid-rise quantization is used without adding the offset of 0.5. This
reduces the signal to noise ratio by approximately 6.02 db, but may be acceptable
for the sake of simplicity when the step size is small.
In digital telephony, two popular quantization schemes are the ’A-law’ (dominant in
Europe) and ‘j-law’(dominant in North America and Japan) .These schemes map
discrete analog values to an 8-bit scale that is nearly linear for small values and then
increases logarithmically as amplitude grows.
Because the human ear’s perception of loudness is roughly logarithmic, this provides
a higher signal to noise ratio over the range of audible sound intensities for a given
number of bits
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Conclusion
During the training at E10B exchange, the whole procedure of call
connecting is well understood. The fact that to connect a call between to end
is a tedious task and it require lot of be performed like first the transmission of
analog signal from source to destination, conversion of analog signal to digital
signal, transmission the signal to the required department. E10B acts as the
local exchange and switching according to time division multiplexing to
connect the call to destination. Lot of precaution are taken in order to provide
smooth, gapless voice transfer between source and destination
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