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Tinniam.V.Ganesh
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Agenda Session 1
1.
2.
3.
4.
5.
6.
7.
8.
9.
10.
11.
12.
13.
14.
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Trends in Telecom
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Technology
Realization
Period
Strowger,
Crossbar
exchanges
Electromechanical
relays. Operator
intervention
~1877 - 1975
Digital
Switches
~1965 onwards
Softswitch
Separation of control
~1996 onwards
and bearer signaling. IP
as transport mechanism
IMS
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~ 2000 onwards
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Historical beginnings of
Telecom
Samuel Morse invents telegraph in 1837.
Alexander Graham Bell invents telephone in 1874.
Marconi experiments with wireless telegraph
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Early switching
exchanges
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Digital Switches
Digital Switches
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Nyquist sampling
frequency
Nyquist sampling frequency requires that in order to faithfully reproduce a
signal at the receiving end the sampling frequency should be twice that of
highest frequency
Voice bandwidth is 300 Hz 4 KHz.
Hence the sampling frequency should be 8 KHz.
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Calculations in
PCM
8 bits per PCM sample and sampled at 8 KHz
8 bits * 8000 samples per sec = 64000 bits/s = 64 Kbps
This is also known as DS0 or E0
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G. 711
G.711 is an ITU-T standard for audio companding. It is primarily used in telephony.
The standard was released for usage in 1972.
G.711 represents logarithmic pulse-code modulation (PCM) samples for signals of
voice frequencies, sampled at the rate of 8000 samples/second.
The -law and A-law algorithms encode 14-bit and 13-bit signed linear PCM
samples (respectively) to logarithmic 8-bit samples. Thus, the G.711 encoder will
create a 64 kbit/s bitstream for a signal sampled at 8 kHz.
There are two main compression algorithms defined in the standard, the -law
algorithm (used in North America & Japan) and A-law algorithm (used in Europe
and the rest of the world).
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Multiplexing
Intelecommunications,multiplexing, is used to refer to aprocesswhere
multiple analog message signals or digital data streams are combined into one
signal over a shared medium. For example, in telecommunications, several
phone calls may be transferred using one wire.
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Time-Division Multiplexing
Ch 1
Ch 2
Ch 3
Ch 4
Ch 5
Ch 6
Ch 7
Ch 8
Time
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Standards in PCM
transmission
There are 2 main standards in the world for transmission of PCM signals
T-Carrier This is used in the North American market
E-Carrier This is used for European market
T1 24 channels * 64 Kbps = 1.544 Mbps
E1 32 channels * 64 Kbps = 2.048 Mbps
Level
North American
European
64 Kbps
64 kbps
8.448 (E2)
34.368 (E3)
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Digital switches
Digital switches aka as Electronic switches receive digitized voice samples.
Voice is sampled and digitized
Digitized PCM voice samples come to the digital switch
Switching or connecting callee and called party happens in memory by
switching the voice samples in two different time slots
Switching happens through a Time-Space-Time switching fabric
Signaling
&
Control
T
D
M
Line
Interfaces
Time
Slot
Interchange
Line
Interfaces
T
D
M
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Time Slot
Interchange
5
k
k
Data in i
Write
Address
Data
out
Read
Address
Speech memory
Connection
memory
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1.
2.
3.
4.
5.
Off-hook or origination
Dialing
Ringing and Ring Back Tone (RBT)
Answer
Disconnection
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Terminations at a Digital
switch
Digital switches connect the following terminations
1.Normal landlines or lines.
2.Trunk Lines (E1,E2,T1,T2 etc)
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1.
Classes of Telecom
switches
Class
5 switch This switch has both lines and trunks. This
switch also supports features like call forwarding, call hold
etc
Class 5 switch
Class 5 switch
Class 4 switch
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Class 5 switch
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Switch B
A dials digits
Initial Address Message (IAM)
Bs phone rings..
B Answers
Answer Message (ANM)
A disconnects
B disconnects
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Wireless
Networks
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Wireless Network
HLR
BSC
SCP
VLR
BTS
Other
MSC
BTS
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24
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WHAT IS COMMUNICATION ?
Communication is used between 2 network elements to exchange information.
There are 2 types of domains in the communication world
Data communication
Telecommunication
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Telecommunication
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Data Communication
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NETWORKS OF TODAY
Data networks are used for making Voice Calls e.g. VOIP
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WHAT IS SIGNALING ?
Signaling refers to the exchange of information between network elements
Signaling between network elements follows a specific protocol
A Protocol refers to the set of rules for communicating between the elements
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SS7 SIGNALING
Signaling happens in a separate channel outside of the voice channels
A separate timeslot is used to transfer signaling messages like call setup,
teardown etc
This is also known as Common Channel Signaling (CCS)
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Switch A
Switch B
Voice Trunk
Signaling Link
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SS7 STACK
CAP,MAP
ISUP,TUP..
Application layer
TCAP
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SCCP
Network layer
MTP 3
Network layer
MTP 2
MTP 1
Physical layer
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SS7 LAYERS
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SS7 PROTOCOLS
SS7 protocols can be classified as call related and non-call related
Call related protocols ISUP (ISDN User Part), B-ISUP (Broadband ISUP)
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The SS7 standard body for North American market is ANSI (American National
Standards Institute). The ANSI versions are ANSI ISUP, ANSI TCAP etc.
The SS7 standard body for European market is ETSI (formerly CCITT). The ETSI versions
of the protocols are ETSI ISUP, ETSI TCAP etc.
There are minor variations in the protocols by the two standard bodies.
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Level
North American
European
64 Kbps
64 kbps
8.448 (E2)
34.368 (E3)
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SS7
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SS7 Stack
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SS7 Layers
Physical Layer (MTP-1)
This defines the physical and electrical characteristics of the signaling links of the SS7
network. Signaling links utilize DS0 channels and carry raw signaling data at a rate of
56 kbps or 64 kbps (56 kbps is the more common implementation).
Message Transfer PartLevel 2 (MTP-2)
The level 2 portion of the message transfer part (MTP Level 2) provides link-layer
functionality. It ensures that the two end points of a signaling link can reliably exchange
signaling messages. It incorporates such capabilities as error checking, flow control,
and sequence checking.
Message Transfer PartLevel 3 (MTP-3)
The level 3 portion of the message transfer part (MTP Level 3) extends the functionality
provided by MTP level 2 to provide network layer functionality. It ensures that messages
can be delivered between signaling points across the SS7 network regardless of whether
they are directly connected. It includes such capabilities as node addressing, routing,
alternate routing, and congestion control.
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SCCP
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The second function provided by the SCCP isGlobal Title translation, the ability to perform
incremental routing using a capability called global title translation (GTT). GTT frees
originating signaling points from the burden of having to know every potential destination to
which they might have to route a message. A switch can originate a query, for example, and
address it to an STP along with a request for GTT. The receiving STP can then examine a
portion of the message, make a determination as to where the message should be routed,
and then route it.
For example, calling-card queries (used to verify that a call can be properly billed to a calling
card) must be routed to an SCP designated by the company that issued the calling card.
Rather than maintaining a nationwide database of where such queries should be routed
(based on the calling-card number), switches generate queries addressed to their local STPs,
which, using GTT, select the correct destination to which the message should be routed.
Note that there is no magic here; STPs must maintain a database that enables them to
determine where a query should be routed. GTT effectively centralizes the problem and
places it in a node (the STP) that has been designed to perform this function.
In performing GTT, an STP does not need to know the exact final destination of a message.
It can, instead, perform intermediate GTT, in which it uses its tables to find another STP
further along the route to the destination. That STP, in turn, can perform final GTT, routing
the message to its actual destination.
Intermediate GTT minimizes the need for STPs to maintain extensive information about
nodes that are far removed from them. GTT also is used at the STP to share load among
mated SCPs in both normal and failure scenarios. In these instances, when messages arrive
at an STP for final GTT and routing to a database, the STP can select from among available
redundant SCPs. It can select an SCP on either a priority basis (referred to as primary
backup) or so as to equalize the load across all available SCPs (referred to as load sharing).
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ISUP
ISDN User Part (ISUP)
ISUP user part defines the messages and protocol used in the establishment and tear
down of voice and data calls over the public switched network (PSN), and to manage the
trunk network on which they rely. Despite its name, ISUP is used for both ISDN and
nonISDN calls. In the North American version of SS7, ISUP messages rely exclusively
on MTP to transport messages between concerned nodes.
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ISUP
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ISUP (ISDN User Part)defines the messages and protocol used in the establishment and
tear down of voice and data calls over the public switched telephone network (PSTN),
and to manage the trunk network on which they rely. Despite its name, ISUP is used for
both ISDN and nonISDN calls. In the North American version of SS7, ISUP messages
rely exclusively on MTP to transport messages between concerned nodes.
ISUP controls the circuits used to carry either voice or data traffic. In addition, the state
of circuits can be verified and managed using ISUP. The management of the circuit
infrastructure can occur both at the individual circuit level and for groups of circuits.
Services that can be defined using ISUP include: Switching, Voice mail, Internet offload.
ISUP is ideal for applications such as switching and voice mail in which calls are routed
between endpoints.
When used in conjunction with TCAP and SIGTRAN, ISUP becomes an enabler for
Internet offload solutions in which Internet sessions of relatively long duration can be
isolated from relatively brief phone conversations.
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IMSI
The IMSI (International Mobile Subscriber Identity) is a unique 15-digit code used to identify an
individual user on a GSM network.
The IMSI consists of three components:
a)
Mobile Country Code (MCC)
b)
Mobile Network Code (MNC)
c)
Mobile Subscriber Identity Number (MSIN)
d)
The IMSI is stored in the Subscriber Identity Module (SIM).
e)
It is also used for acquiring other details of the mobile in theHome Location
Register(HLR) or as locally copied in theVisitor Location Register.
f)
The IMSI is used inanymobile network that interconnects with other networks, in
particularCDMAandEVDOnetworks as well as GSM networks. This number is
provisioned in the phone directly
IMSI - 310150123456789
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MCC
310
USA
MNC
150
AT&T
MSIN
123456789
MSIN
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TMSI
TMSI Temporary Mobile Subscriber Identity
A TMSI is used to protect the true identity (IMSI) of a subscriber. It is issued by and
stored within a VLR (not in the HLR) when an IMSI attach takes place or a Location Area
(LA) update takes place. At the MS it is stored in the MSs SIM. The issued TMSI only
has validity within a specific LA.
Since TMSI has local significance, the structure may be chosen by the administration. It
should not be more than four octets.
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MSISDN
MSISDN
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IMEI
IMEI
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MSRN
MSRN Mobile Station Roaming Number
The MSRN is a temporary, location-dependant ISDN number issued by the parent VLR to all
MSs within its area of responsibility. It is stored in the VLR and associated HLR but not
in the MS. The MSRN is used by the VLR associated MSC for call routing within the
MSC/VLR service area.
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Intelligent Network
GMSC
gsmSSF
C
gsmSRF
gsmSCF
HLR
gsmSSF VLR
MSC
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INAP
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Intelligent Network Application Part (INAP) is the signaling protocol used in Intelligent
Networking. Developed by the International Telecommunication Union(ITU), IN is
recognized as a global standard. Within the International Telecommunications Union, a
total functionality of the IN has been defined and implemented in digestible segments
called capability sets. The first version to be released was Capability Set 1 (CS-1).
Currently CS-2 is defined and available. The CAMEL Application Part (CAP) is a
derivative of INAP and enables the use of INAP in mobile GSM networks.
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Service Switching Point (SSP)is a physical entity in the Intelligent Network that
provides the switching functionality. SSP the point of subscription for the service user,
and is responsible for detecting special conditions during call processing that cause a
query for instructions to be issued to the SCP.
The SSP contains Detection Capability to detect requests for IN services. It also contains
capabilities to communicate with other physical entities containing SCF, such as SCP,
and to respond to instructions from the other physical entities. Functionally, an SSP
contains a Call Control Function, a Service Switching Function, and, if the SSP is a
local exchange, a Call Control Agent Function. It also may optionally contain Service
Control Function, and/or a Specialized Resource Function, and/or a Service Data
Function. The SSP may provide IN services to users connected to subtending Network
Access Points.
The SSP is usually provided by the traditional switch manufacturers. These switches are
programmable and they can be implemented using multipurpose processors. The main
difference of SSP from an ordinary switch is in the software where the service control of
IN is separated from the basic call control.
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Service Control Point (SCP)validates and authenticates information from the service
user, processing requests from the SSP and issuing responses.The SCP stores the
service provider instructions and data that direct switch processing and provide call
control. At predefined points during processing an incoming or outgoing call, the switch
suspends what it is doing, packages up information it has regarding the processing of
the call, and queries the SCP for further instruction. The SCP executes user-defined
programs that analyze the current state of the call and the information received from
the switch. The programs can then modify or create the call data that is sent back to
the switch. The switch then analyzes the information received from the SCP and follows
the provided instruction to further process the call.
Functionally, an SCP contains Service Control Function (SCF) and optionally also
Service Data Function (SDF). The SCF is implemented in Service Logic Programs (SLP).
The SCP is connected to SSPs by a signalling network. Multiple SCPs may contain the
same SLPs and data to improve service reliability and to facilitate load sharing between
SCPs. I
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Switch B
A dials digits
Initial Address Message (IAM)
Bs phone rings..
B Answers
Answer Message (ANM)
A disconnects
B disconnects
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ISUP
A simple call flow using ISUP signaling is as follows:
Call set up:When a call is placed to an out-of-switch number, the originating SSP transmits an
ISUP initial address message (IAM) to reserve an idle trunk circuit from the originating
switch to the destination switch. The destination switch rings the called party line if the line
is available and transmits an ISUP address complete message (ACM) to the originating
switch to indicate that the remote end of the trunk circuit has been reserved. The STP
routes the ACM to the originating switch which rings the calling party's line and connects it
to the trunk to complete the voice circuit from the calling party to the called party.
Call connection:When the called party picks up the phone, the destination switch terminates
the ringing tone and transmits an ISUP answer message (ANM) to the originating switch via
its home STP. The STP routes the ANM to the originating switch which verifies that the
calling party's line is connected to the reserved trunk and, if so, initiates billing.
Call tear down:If the calling party hangs-up first, the originating switch sends an ISUP release
message (REL) to release the trunk circuit between the switches. The STP routes the REL to
the destination switch. If the called party hangs up first, or if the line is busy, the
destination switch sends an REL to the originating switch indicating the release cause (e.g.,
normal release or busy). Upon receiving the REL, the destination switch disconnects the
trunk from the called party's line, sets the trunk state to idle, and transmits an ISUP
release complete message (RLC) to the originating switch to acknowledge the release of the
remote end of the trunk circuit. When the originating switch receives (or generates) the RLC,
it terminates the billing cycle and sets the trunk state to idle in preparation for the next call.
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Questions ?
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Quiz 1
1.
2.
3.
4.
5.
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PCM is the technique where the magnitude of the signal is sampled and digitized.
a) True
b) False
The bit rate of DS0 or E0 PCM channel is
a) 56 Kbps b) 2 Mbps
c) 64 Kbps
d) 8 Khz
Time division multiplexing is
a. Transferring multiple lower rate channels onto higher bit rate channel
b. Sampling a signal in multiples of time
c. Using several carrier frequencies to multiplex a channel
d. Uses G.711 law
T Carrier is North American market and E Carrier is European
a. True
b. False
The principle of digital switching is based on
a. Mapping IP addresses to port numbers
b. Performing layer 2 switching
c. Based on Time slot interchange
d. OSI Network layer
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Quiz 1
6.
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Agenda Session 2
Evolution of Wireless Technologies
1. Comparison of SS7 & OSI stack
2. 1G
3. 2G
4. 2.5G
5. 3G
6. 3.5G
7. 4G
8. Recap
9. TDMA, FDMA, CDMA
10. CDMA Basics
11. GSM Architecture
12. Access, Core Network
13. Cellular concepts - Roaming, Registration, Handoff
14. Traffic Engineering concepts
15. Recap
16. Quiz 2
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Similar to AMPS,
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Digital Systems
Leverage technology to increase capacity
Speech compression, digital signal processing
Greater security against fraud
Variety of 2G Systems
IS-54 and IS-136
Uses Time Division Multiplexing (TDM). Introduced in 1990 in North America
Digital voice channels and analog control channels
IS-136
Introduced in 1994 in North America
Digital Voice and digital control channels
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HLR
VLR
BSC
Abis
MSC
A
MS
AuC
GMSC
BTS
E
Abis
A
MSC
F
EIR
E
PSTN
BSC
Um
BTS
X.25
VLR
X.25
OMC Server
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GPRS (2.5 G)
GPRS
GPRS is an enhancement over the GSM and adds some nodes in the network to provide the
packet switched services. These network nodes are called GSNs (GPRS Support Nodes)
and are responsible for the routing and delivery of the data packets to and from the MS
and external packet data networks (PDN).
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3G Rel 99 Architecture
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Benefits of 3G
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Fourth Generation 4G
To handle even higher data throughputs we have the 4G technology
1.
Long Term Evolution (LTE)
2.
Wireless Interoperability for Microwave Access (WiMAX)
3.
Uses an all-IP core network
4.
Data rates upto 100 Mbps
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1G Technologies'
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FDMA
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2G - TDMA
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TDMA
TDMA is an assigned frequency band shared among a few users. However, each user is
allowed to transmit in predetermined time slots. Hence, channelization of users in the
same band is achieved through separation in time.
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Frequency Division Multiple Access allocates a discrete amount of bandwidth per user
Time Division Multiple Access allocates unique time slots for each user
Code Division Multiple Access all users share the same frequency all the time. A
unique code assigned to each user allows it to be distinguished from other users
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Introduction to CDMA
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Fc carrier
bandwidth
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Amplitude modulation : The bandwidth is twice the baseband on either side of the
carrier
Frequency modulation modulates the carrier frequency with the baseband signal
Digital modulation like QPSK give higher spectral efficiency
In spread spectrum the transmitted signal is spread using a bandwidth much larger
than that required by mixing the data and the spreading code signal.
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Spread information
110101110100 110101110100
Cyclic code
generator
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Direct Sequence (DS) carrier modulated by a digital code larger than the signal
information bit rate. These systems are also called Pseudo-noise systems
Time Hopping (TH) transmission time divided into frames and frames into time
slots. During each frame one and only one time slot is modulated with the
message.
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97
Baseband
modulation
Baseband de modulation
PN sequence
generator
Information
bits
PN Sequence
generator
* chip each bit in a PN sequence is called chip to distinguish it from information bits
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Frequency Hopping
Time
f1
f2
f3
f4
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Frequency
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99
Frequency Hopping
Information
bits
Baseband
modulation
PN sequence
generator
Bandpass
filter
Frequency
synthesizer
Baseban
de modulati
on
Frequency
synthesizer
Information
bits
PN Sequence
generator
On the reception side the PN sequence generator defines the centre frequency of
bandpass filter and the frequency for the demodulation process. The demodulation can
only succeed if both the transmission and reception are synchronized.
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100
CDMA
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101
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Background to GSM
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103
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104
INK
L
N
W
DO
890MHz
915MHz
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935MHz
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960MHz
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GSM Architecture
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107
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Network Subsystem
Can be considered as a heart of the GSM Network.
All the major activities like
Routing,
Security functions,
Call handling, charging,
Operation & maintenance,
Handover decisions,
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Various kinds of interfaces are used to communicate between the different entities.
Different methods are used to optimize and provide the quality network with the
minimum operating cost.
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110
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2G Architecture
ISUP
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2G Architecture
Gi
PSTN
GMSC
GGSN
AuC
C
PSTN
HLR
PSTN
VLR
Gf
Gs
MSC
Gn
Gr
EIR
F
Gc
VLR
SGSN
MSC
Gp
CN
A
Gb
IuPS
IuCS
RNS
BSS
BSC
Abis
BTS
Iur
RNC
RNC
Iubis
BTS
NodeB
NodeB
cell
Um
Uu
ME
SIMMEi/f
SIM
or
Cu
USIM
MS
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Registration
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Every mobile in the network communicates its location, and identification to the
network through the registration process
Knowing the location allows the BTS to page the mobile when a MS terminated
call is requested.
Power up registration occurs when the MS is turned on and enter the mobile
idle state
Power down registration when turned off
Timer based registration: The MS must register according to preprogrammed timer
Distance based registration : When it reaches a pre-specified distance from
the BTS
Zone based registration: occurs based on internal zone configuration ,
when a MS enter a new zone
Parameter change registration: Occurs when a parameter changes
Ordered registration : occurs every time the system requests registration
Traffic channel registration: occurs when the MS registers while requesting
a traffic channel allocation
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Roaming
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MSs are considered home when they are located in their home system, where they are
registered and allowed to operate
They are roamers when they are out of their home systems
MSs have a list of locations where they are in the home system
Roamingis a general term referring to the extension of connectivity service in a location
that is different from the home location where the service was registered.
the ability for acellularcustomer to automatically make and receive voice calls, send
and receive data, or access other services, including home data services, when travelling
outside the geographical coverage area of the homenetwork, by means of using a visited
network
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Handoffs
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BS Traffic not balanced: Network monitors traffic and trigger handoffs if load
not balanced among BS
Distance limit exceeded
Pilot signal strength below threshold the MS can initiate a handoff
Power level exceeded When the mobile has exceeded the power threshold
then either side can initiate a handoff
Handoff involves the mobile moving to a new traffic channel of a different BS
Types of handoff
Soft Handoff MS has simultaneous connections with two BS before a
decision is made as to which signal is stronger before breaking the
connection with the BS with the weaker signal strength (not perceived by
the user)
Hard Handoff There is a break before the make. Connection to old
traffic channel is broken before the connection to a new one is made
(user hears a click)
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Authentication
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The AUC does not engage directly in the authentication process, but instead generates
data known astripletsfor theMSCto use during the procedure. The security of the
process depends upon a shared secret between the AUC and the SIM called theKi. The
Ki is securely burned into the SIM during manufacture and is also securely replicated
onto the AUC. This Ki is never transmitted between the AUC and SIM, but is combined
with theIMSIto produce a challenge/response for identification purposes and
anencryption keycalledKcfor use in over the air communications.
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Authentication procedure
1. Algorithm id(the standard algorithms are called A3 or A8, but an operator may choose a
proprietary one). When the MSC asks the AUC for a new set of triplets for a particular
IMSI, the AUC first generates a random number known asRAND. This RAND is then
combined with the Ki to produce two numbers as follows:
2. The Ki and RAND are fed into the A3/A8 (or other operator proprietary algorithm) and a
number known as Signed RESponse orSRESis calculated.
3. The Ki and RAND are fed into a standard A5 algorithm and a number called the Kc is
calculated.
4. The numbers (RAND, SRES, KC) form the triplet sent back to the MSC. When a
particularIMSIrequests access to the GSM core network, the MSC sends theRANDpart
of the triplet to the SIM. The SIM then feeds this number and the Ki (which is burned
onto the SIM) into the A3/A8/proprietary algorithm as appropriate and an SRES is
calculated and sent back to the MSC. If this SRES matches with the SRES in the triplet
(which it should if it is a valid SIM), then the mobile is allowed to attach and proceed
withGSM services.
5. After successful authentication, the MSC sends the encryption key Kc to theBase Station
Controller(BSC) so that all communications can be encrypted and decrypted. Of course,
the mobile phone can generate the Kc itself by feeding the same RAND supplied during
authentication and the Ki into the A5 algorithm.
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Authentication
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Mobility Management
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Traffic Engineering
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Traffic Engineering
Trunk
the telephone lines connecting one telephone switch or exchange with
another are called trunks.
Calling rate (C)
The number of calls which arrive over a time interval
Holding time (H)
The average duration of a call. The duartion the telephony circuits are held
during conversation
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Traffic Calculations
The erlang describes the total traffic volume of one hour, or 3600 seconds.
The traffic intensity, more often called the traffic, is defined as the average number of
calls in progress.
A = C x H/T
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Traffic Problem
On average, during the busy hour, a company makes 120 outgoing calls of average duration
2 minutes. It receives 200 incoming calls of average duration 3 minutes. Find the
outgoing traffic, the incoming traffic and the total traffic.
A = C x H /T
Solution
where T = 1 hour = 60 minutes
Outgoing traffic = 120 calls x 2 minutes/ 60 minutes = 4 E
Incoming traffic = 200 calls x 3 minutes/ 60 minutes =10 E
Total traffic = 4 E + 10 E = 14 E
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Traffic terms
Lost call or blocked calls
In a circuit-switched system, all attempts to make calls over a congested group of trunks are
unsuccessful. The unsuccessful call is called lost call or blocked call.
Grade of service
probability of meeting blockage is called the grade of service (B)
Example: On average, one call in 100 will be blocked
B= 1/100 = 0.01
Grade of service is also the
proportion of the time for which congestion exists
probability of congestion
probability that a call will be lost due to congestion
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Traffic calculations
Example
During the busy hour, 1200 calls were offered to a group of trunks and six calls were lost.
The average call duration was 3 minutes
The traffic offered = A = C1 x H/T = 1200 x 3 /60 = 60 E
The traffic carried = C2 x H/T=(1200-6) x 3 / 60 = 59.7 E
The traffic lost = B = C3 x H/T = 6 x 3 / 60 = 0.3 E
Grade of service = B/A = 0.3 / 60 = 0.005
The total duration of the periods of congestion = B x T = 0.005 x 3600 =18 seconds
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Wireless Network
MS
Um
BTS
HLR
VLR
BSC
Abis
MSC
A
MS
AuC
GMSC
BTS
E
Abis
A
MSC
F
EIR
E
PSTN
BSC
Um
BTS
X.25
VLR
X.25
OMC Server
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Questions ?
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Quiz 2
1.
2.
3.
4.
5.
6.
7.
8.
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Quiz 2
9.
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Tinniam V Ganesh
tvganesh.85@gmail.com
Read my blogs: http://gigadom.wordpress.com/
http://savvydom.wordpress.com/
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