A

REPORT

ON

PRACTICAL TRAINING

AT

“TATA TELESERVICE LIMITED”

JAIPUR,RAJASTHAN

SESSION: 2010-11

Submitted By:-
KULDEEP JAIN
B.TECH. VII SEM(IV YEAR)

Department Of Electronics &Communication

Engineering

MAHARISHI ARVIND INSTITUTE OF ENGINEERING

& TECHNOLOGY,JAIPUR

(AFFILIATED BY RAJASTHAN TECHNICAL UNIVERSITY,KOTA)

 ACKNOWLEDGEMENT

I wish to thank all those who had a major influence on the conception and
fulfillment of this project report. This report has been written in context of the
fulfillment of the industrial training, after my third year, under the guidance of Mr.
Rahul Sharma, who gave me all the help and support I needed and also motivated
me to work hard during the course of this project.

I would like to thank Mr. Dharmendra Sharma(General Manager, Tata

Teleservices), for giving me the opportunity to work with his team.

I also extend to all the engineers in the Technology department who always
guided, helped and supported me.

Kuldeep Jain
Electronics and Communication
THE TATA GROUP
Vision

To set the standards in their chosen businesses and markets .Tata group will be an agile,
customer-driven company that earns trust through responsible leadership, innovation and
excellence. They are committed to provide a caring, empowered and fun filled environment.

Mission

They are going to serve the fixed and mobile voice and data communication needs of business
and individual customers by offering reliable and responsive services of the highest standard in
consonance with the TATA reputation.
The purpose in Tata is to improve the Quality of Life in India through leadership in sectors of
National Economic Significance to which group bring a unique set of capabilities. This requires
group to grow aggressively in focused areas of business.
Tata’s heritage of returning to society what they earn evokes trust among consumers, employees,
shareholders and the community. This heritage will be continuously enriched by formalizing the
high standards of behavior expected from employees and companies.
The Tata name is a unique asset representing leadership with trust. Leveraging this asset to
enhance group synergy and become globally competitive is the route to sustain growth and long-
term success.
Core Values
The Tata Group has always sought to be a value-driven organization. These values continue to
direct the group's growth and businesses. The five core Tata values underpinning the way we do
business are:
• Integrity - must conduct our business fairly, with honesty and transparency. Everything
we do must stand the test of public scrutiny.
• Understanding - must be caring, show respect, compassion and humanity for our
colleagues and customers around the world and always work for the benefit of India.
• Excellence - must constantly strive to achieve the highest possible standards in our day-
to-day work and in the quality of the goods and services we provide.
• Unity - must work cohesively with our colleagues across the group and with our
customers and partners around the world, building strong relationships based on
tolerance, understanding and mutual cooperation.
• Responsibility – must continue to be responsible, sensitive to the countries, communities
and environments in which it works, always ensuring that what comes from the people
goes back to the people many times over.
Company Profile
The telecommunication revolution is sweeping India today, and Tata Teleservices Limited
(TTSL) is spearheading this change with its range of integrated telecommunication services.
Incorporated in 1996, Tata Teleservices was the first to launch CDMA mobile services in India
with the Andhra Pradesh circle. The company offers services under the brand name ‘Tata
Indicom’ in eight key Indian circles of Andhra Pradesh, Delhi, Gujarat, Karnataka, Maharashtra,
Mumbai, Tamil Nadu and Chennai comprising 70% of the telecom revenue potential of the
country.

Starting with the major acquisition of Hughes Telecom (India) limited [now renamed Tata tele
services (Maharashtra) Limited] in December 2002, the company has swung in to expansion
mode. The company has recently acquired a universal Access Services License (UASL) for 11
new circles. The new circles are Bihar, Haryana, Himachal Pradesh, Kerala, Kolkata, Orissa,
Punjab, Rajasthan, Uttar Pradesh (East), Uttar Pradesh (west) and west Bengal.
Critical to its success is the application of cutting-edge through collaborations with leading firms
like Lucent Technologies, Motorola, Schlumberger, Kenan and Oracle
The TTSL has established a strong foothold in the Maharashtra Circle, by acquiring 71% stake in
Hughes Tele.com (India) Ltd (HTIL). The company, which has been renamed Tata Teleservices
(Maharashtra) Ltd., has close to 2,00,000 subscribers in the state with more than 1,20,000
subscribers in Mumbai alone.
The company pioneered the CDMA 3G1X technology platform in India, and has established a
robust and reliable telecom infrastructure that ensures quality in its services. Tata teleservices has
partnered Motorola, Ericsson, lucent, GTL, ZTE and ECI telecom for the deployment of a
reliable, technologically advanced network.

Various Departments at TTSL, Jaipur

Various department at TTSL, Jaipur (Raj.) functioning exclusively in a defined way for
achieving its objectives

CUSTOMER CARE HUMAN RESOURCE

FINANCE ISIT

BILLING SALES

MARKETING OPERATION AND SUPPORT

NETWORKS
NETWORK
If there were only three or four telephones in a locale, it would make sense to connect each
phone to all other phones and find a simple method of selecting the desired one. However, if
there are three or four thousand phones in a locale, such a method is not useful. Then it is
appropriate to connect each phone to some centrally located office and perform switching there.
This switching could be a simple manual operation using plugs and sockets or could be done
with electromechanical devices or with electronics. In any case, this “central office” solution is
the one that has been chosen by the telecommunications industry. As we connect each of these
thousands of telephones to the central office, we have what is called a star configuration; all lines
are particular to one and only one station, and all terminate on the nucleus of this star—the
central office (CO). These connections are called the local exchange plant, and the telephone
company handling this function is called the local exchange carrier (LEC). The connections
themselves are often called the “local loop”. In more technical terms, the section closest to the
customer’s premises is called the distribution plant and that section closest to the central office,
the feeder plant.
To connect one city or state or country to another, these central offices are to be connected to
higher echelon central offices. This forms a number of levels of central offices known as
hierarchy of switching on which actually the telecommunication industry depends.

The Hierarchy of Switching Systems in Its

Most Basic Form Consists of Five “Classes” of Offices

Here, the only office that has consumers as its subscribers is the Class-5 office. The other offices
in this hierarchy have lower-level central offices as their “subscribers.” Those lines connecting
switching offices to switching offices, rather than to subscribers, are called trunks. The section
leading upward from the Class-5 offices is handled not by the LECs, but by the interexchange
carriers (IXCs), the long-distance carriers. The total network is called the public switched
telephone network (PSTN). The interconnections among the various COs can be twisted copper
pair carrier systems utilizing copper pairs (e.g., T1), microwave, satellites, and certainly fiber.

MOBILE COMMUNICATION

A mobile phone or mobile is a long-range, electronic device used for mobile voice or data
communication over a network of specialized base stations known as cell sites. A mobile phone,
offer full duplex-communication and automates calling to and paging from a public switched
telephone network (PSTN), handoff or handover during a phone call when the user moves from
one cell (base station coverage area) to another. It operates on the principle of wireless
communication.
Each mobile uses a separate, temporary radio channel to talk to the cell site. A cell is the basic
geographic unit of a cellular system. Cells are base stations transmitting over small geographic
areas that are represented as hexagons. Each cell size varies depending on the landscape.
The cell site talks to many mobiles at once, using one channel per mobile. Channels use a pair of
frequencies for communication—one for transmitting from the cell site, the forward link, and
one frequency for the cell site to receive calls from the users, the reverse link. Radio energy
dissipates over distance, so mobiles must stay near the base station to maintain communications.
The basic structure of mobile networks includes telephone systems and radio services. While a
mobile radio service operates in a closed network and has no access to telephone systems, mobile
telephone service allows interconnection to a telephone network.

Basic Mobile Telephone Service Network

Earlier there was only one powerful transmitter located at the highest spot in an area which
would be able to broadcast in a radius of up to 50 kilometers. This resulted in interference
problems caused by mobile units due to the use of same channel in adjacent areas and hence
reduction in efficiency.

Thus, the cellular concept came into picture which states that instead of using one powerful
transmitter, many low-power transmitters were placed throughout a coverage area, which allows
cells to be sized according to the subscriber density and demand of a given area. Hence, as the
population grows, cells can be added to accommodate that growth. Frequencies used in one cell
cluster can be reused in other cells. Conversations can be handed off from cell to cell to maintain
constant phone service as the user moves between cells. The channel is made of two frequencies,
one for transmitting to the base station and one to receive information from the base station.

Mobile Telephone System Using a Cellular Architecture

NETWORK COMPONENTS

There are a number of components which constitute the whole architecture of a network. Before
going into the details of network architecture, there are some transmission concepts widely
being used in the network.

E-carrier
In digital telecommunications, where a single physical wire pair can be used to carry many
simultaneous voice conversations, worldwide standards have been created and deployed. The
European Conference of Postal and Telecommunications Administrations (CEPT) originally
standardized the E-carrier system, which is now widely used in almost all countries.
The E-carrier standards form part of the Plesiochronous Digital Hierarchy (PDH) where
groups of E1 circuits may be bundled onto higher capacity E3 links between telephone
exchanges or countries. This allows a network operator to provide a private end-to-end E1 circuit
between customers in different countries that share single high capacity links in between.
In practice, only E1 (30 circuit) and E3 (480 circuit) versions are used. Unlike Internet data
services, E-carrier systems permanently allocate capacity for a voice call for its entire duration.
This ensures high call quality because the transmission arrives with the same short delay
(Latency) and capacity at all times.
E1 circuits are very common in most telephone exchanges and are used to connect to medium
and large companies, to remote exchanges and in many cases between exchanges. E3 lines are
used between exchanges, operators and/or countries, and have a transmission speed of 34.368
Mbit/s.
An E1 link operates over two separate sets of wires, usually twisted pair cable. A nominal 3 Volt
peak signal is encoded with pulses using a method that avoids long periods without polarity
changes. The line data rate is 2.048 Mbit/s (full duplex, i.e. 2.048 Mbit/s downstream and 2.048
Mbit/s upstream) which is split into 32 timeslots, each being allocated 8 bits in turn. Thus each
timeslot sends and receives an 8-bit sample 8000 times per second (8 x 8000 x 32 = 2,048,000).
This is ideal for voice telephone calls where the voice is sampled into an 8 bit number at that
data rate and reconstructed at the other end. The timeslots are numbered from 0 to 31.
One timeslot (TS0) is reserved for framing purposes, and alternately transmits a fixed pattern.
This allows the receiver to lock onto the start of each frame and match up each channel in turn.
The standards allow for a full Cyclic Redundancy Check to be performed across all bits
transmitted in each frame, to detect if the circuit is losing bits (information), but this is not
always used.
One timeslot (TS16) is often reserved for signaling purposes, to control call setup and teardown
according to one of several standard telecommunications protocols. This includes Channel
Associated Signaling (CAS) where a set of bits is used to replicate opening and closing the
circuit or using tone signaling which is passed through on the voice circuits themselves.
The STM-1 (Synchronous Transport Module) is the SDH ITU-T fiber optic network
transmission standard. It has a bit rate of 155.52 Mbit/s. It is another technology for E-carrier.
STM-1 carries 63 E1’s traffic. The other levels are STM-4, STM-16 and STM-64.

Signaling System Number 7 (SS7)

Signaling System Number 7 (SS7) is a set of telephony signaling protocols which are used to
set up most of the world's public switched telephone network telephone calls. The main purpose
is to set up and tear down telephone calls. Other uses include number translation, prepaid billing
mechanisms, short message service (SMS), and a variety of other mass market services.
SS7 moved to a system in which the signaling information was out-of-band, carried in a separate
signaling channel. This avoided the security problems earlier systems had, as the end user had no
connection to these channels. SS7 is referred to as so-called Common Channel Interoffice
Signaling Systems (CCIS) or Common Channel Signaling (CCS) due to their hard separation
of signaling and bearer channels. This required a separate channel dedicated solely to signaling,
but the greater speed of signaling decreased the holding time of the bearer channels, and the
number of available channels was rapidly increasing anyway at the time SS7 was implemented.
When the signaling is performed on the same circuit that will ultimately carry the conversation
of the call, it is termed Circuit-Associated Signaling (CAS).
In stark contrast, SS7 signaling is termed Non-Circuit-Associated Signaling (NCAS) 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 Non-Circuit-Associated
Signaling, it becomes possible to exchange signaling without first seizing a facility, leading to
significant savings and performance increases in both signaling and facility usage.
SS7 is designed to operate in two modes:

• Associated Mode
• Quasi-Associated Mode
When operating in the Associated Mode, SS7 signaling progresses from switch to switch through
the PSTN following the same path as the associated facilities that carry the telephone call. This
mode is more economical for small networks.
When operating in the Quasi-Associated Mode, SS7 signaling progresses from the originating
switch to the terminating switch following a path through a separate SS7 signaling network
composed of STPs. This mode is more economical for large networks with lightly loaded
signaling links.
SS7 clearly splits the signaling planes and voice circuits. An SS7 network has to be made up of
SS7-capable equipment from end to end in order to provide its full functionality. The network is
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 point code. Extended services are
provided by a database interface at the SCP level using the SS7 network.
Some of the services includes: call forwarding (busy and no answer), voice mail, call waiting,
conference calling, calling name and number display, called name and number display, call
screening, malicious caller identification, busy callback.

ARCHITECTURE

The network is divided into following sections:-

Mobile Station (MS)

It consists of the terminal (TE) and a smart card called the subscriber identity module (SIM).
The SIM provides personal mobility so that user can have access to subscribed services
irrespective of a specific terminal. The SIM card is the actual place where the network finds the
telephone number of the user. Thus by inserting the SIM card into another terminal, the user is
able to use the new terminal receive, make calls and use other subscribed services while using
the same telephone number.
The network terminal is uniquely identified by the International Mobile Equipment Identity
(IMEI). The SIM card contains the International Mobile Subscriber Identity (IMSI) used to
identify the subscriber to the system, a secret key for authentication and other information.

The structures of IMEI and IMSI are shown below:

The IMEI can be up to 15 digits and comprises:

• 3-digit Type Approval Code (TAC). This is given to the unit after it passes conformance
tests.

• 1 or 2 digit Final Assy Code (FAC). This identifies the place of manufacture of assembly
of the MS unit

• 11 digit MS unit serial number

• 1 spare digit reserved for future use.

The IMSI is also 15 digits and consists of:

• 3-digit Mobile Country Code (MCC). This identifies the country where the system
operates.

• 2-digit Mobile Network Code (MNC). This uniquely identifies each cellular provider.

• The Mobile Subscriber Identification Code (MSIC). This uniquely identifies each
customer of the provider.

In CDMA technology, the mobile station is contained in the handset only which is carried by the
subscriber.

Base Station Subsystem:-

The base station subsystem (BSS) is the section of a traditional cellular telephone network
which is responsible for handling traffic and signaling between a mobile phone and the network
switching subsystem. The BSS carries out trans-coding of speech channels, allocation of radio
channels to mobile phones, paging, quality management of transmission and reception over the
air interface and many other tasks related to the radio network.
The Base Station Subsystem is composed of two parts:-
The Base Transceiver Station (BTS) and the Base Station Controller (BSC).

Base Transceiver Station (BTS)

The base transceiver station, or BTS, contains the equipment for transmitting and receiving of
radio signals (transceivers), antennas, and equipment for encrypting and decrypting
communications with the base station controller (BSC). Typically a BTS for anything other than
a Pico-cell will have several transceivers (TRXs) which allow it to serve several different
frequencies and different sectors of the cell (in the case of sectorized base stations). A BTS is
controlled by a parent BSC via the base station control function (BCF). The BCF is implemented
as a discrete unit or even incorporated in a TRX in compact base stations. The BCF provides an
operations and maintenance (O&M) connection to the network management system (NMS), and
manages operational states of each TRX, as well as software handling and alarm collection.
The functions of a BTS vary depending on the cellular technology used and the cellular
telephone provider. There are vendors in which the BTS is a plain transceiver which receives
information from the MS (mobile station) through the Um (air interface) and then converts it to a
TDM ("PCM") based interface, the Abis interface, and sends it towards the BSC. There are
vendors which build their BTSs so the information is preprocessed, target cell lists are generated
and even intra-cell handover (HO) can be fully handled. The advantage in this case is fewer
loads on the expensive Abis interface.
Antenna combiners are implemented to use the same antenna for several TRXs (carriers), the
more TRXs are combined the greater the combiner loss will be. Up to 8:1 combiners are found in
micro and Pico cells only.
Frequency hopping is often used to increase overall BTS performance; this involves the rapid
switching of voice traffic between TRXs in a sector. A hopping sequence is followed by the
TRXs and handsets using the sector. Several hopping sequences are available, and the sequence
in use for a particular cell is continually broadcast by that cell so that it is known to the handsets.
A TRX may lose some of this capacity as some information is required to be broadcast to
handsets in the area that the BTS serves. This information allows the handsets to identify the
network and gain access to it. This signaling makes use of a channel known as the broadcast
control channel (BCCH).

Sectorization
By using directional antennae on a base station, each pointing in different directions, it is
possible to sectorize the base station so that several different cells are served from the same
location. Typically these directional antennas have a beam-width of 65 to 85 degrees. This
increases the traffic capacity of the base station (each frequency can carry eight voice channels)
whilst not greatly increasing the interference caused to neighboring cells as in any given
direction, only a small number of frequencies are being broadcast. Typically two antennas are
used per sector, at spacing of ten or more wavelengths apart. This allows the operator to
overcome the effects of fading due to physical phenomena such as multipath reception. Some
amplification of the received signal as it leaves the antenna is often used to preserve the balance
between uplink and downlink signal.

Base station controller (BSC)

The base station controller (BSC) provides, classically, the intelligence behind the BTS’s.
Typically a BSC has tens or even hundreds of BTSs under its control. The BSC handles
allocation of radio channels, receives measurements from the mobile phones, and controls
handovers from BTS to BTS. A key function of the BSC is to act as a concentrator where many
different low capacity connections to BTSs (with relatively low utilization) become reduced to a
smaller number of connections towards the mobile switching center (MSC) (with a high level
of utilization). Overall, this means that networks are often structured to have many BSCs
distributed into regions near their BTSs which are then connected to large centralized MSC sites.
The BSC is undoubtedly the most robust element in the BSS as it is not only a BTS controller
but, for some vendors, a full switching center, as well as an SS7 node with connections to the
MSC and serving GPRS support node (SGSN) (when using GPRS). It also provides all the
required data to the operation support subsystem (OSS) as well as to the performance measuring
centers.
A BSC is often based on a distributed computing architecture, with redundancy applied to
critical functional units to ensure availability in the event of fault conditions. The databases for
all the sites, including information such as carrier frequencies, frequency hopping lists, power
reduction levels, receiving levels for cell border calculation, are stored in the BSC. This data is
obtained directly from radio planning engineering which involves modeling of the signal
propagation as well as traffic projections.

BSS interfaces

Um
It is the air interface between the mobile station (MS) and the BTS. This interface uses
LAPDm protocol for signaling, to conduct call control, measurement reporting, handover,
power control, authentication, authorization, location update and so on. Traffic and
signaling are sent in bursts of 0.577 ms at intervals of 4.615 ms, to form data blocks each
20 ms.
Abis
It is the interface between the BTS and BSC. It is generally carried by a DS-1, ES-1, or
E1 TDM circuit. Uses TDM sub-channels for traffic (TCH), LAPD protocol for BTS
supervision and telecom signaling, and carries synchronization from the BSC to the BTS
and MS.
A9/A10
It is the interface between the BSC and MSC. It is used for carrying traffic channels and
the BSSAP user part of the SS7 stack. Although there are usually transcoding units
between BSC and MSC, the signaling communication takes place between these two
ending points and the transcoder unit doesn't touch the SS7 information, only the voice or
CS data are transcoded.
Gb
It connects the BSS to the SGSN in the GPRS core network.

NETWORK SWITCHING SUBSYSTEM (NSS):-

Network switching subsystem (NSS) is the component of a network that carries out switching
functions and manages the communications between mobile phones and the Public Switched
Telephone Network (PSTN). It is owned and deployed by mobile phone operators and allows
mobile phones to communicate with each other and telephones in the wider telecommunications
network. The architecture closely resembles a telephone exchange, but there are additional
functions which are needed because the phones are not fixed in one location. Each of these
functions handles different aspects of mobility management.
The Network Switching Sub system usually refers to the circuit-switched core network, used for
traditional services such as voice calls, SMS, and circuit switched data calls. Mobile phones also
have access to services such as WAP, MMS, and Internet access.

Mobile switching center (MSC)

The mobile switching center (MSC) is responsible for handling voice calls and SMS as well as
other services such as conference calls, FAX and circuit switched data. The MSC sets up and
releases the end-to-end connection, handles mobility and hand-over requirements during the call
and take care of charging and real time pre-paid account monitoring.
In the GSM mobile phone system, in contrast with earlier analogue services, fax and data
information is sent directly digitally encoded to the MSC. Only at the MSC is this re-coded into
an "analogue" signal as PCM signal in a 64-kbit/s timeslot.
There are various different names for MSCs in different contexts which reflect their complex
role in the network; all of these terms though could refer to the same MSC, but doing different
things at different times.
The gateway MSC (G-MSC) is the MSC that determines which visited MSC the subscriber who
is being called is currently located. It also interfaces with the PSTN. All mobile to mobile calls
and PSTN to mobile calls are routed through a G-MSC. The term is only valid in the context of
one call since any MSC may provide both the gateway function and the Visited MSC function;
however, some manufacturers design dedicated high capacity MSCs which do not have any
BSSs connected to them. These MSCs will then be the Gateway MSC for many of the calls they
handle.
The visited MSC (V-MSC) is the MSC where a customer is currently located. The VLR
associated with this MSC will have the subscriber's data in it.
The anchor MSC is the MSC from which a handover has been initiated.
The target MSC is the MSC toward which a Handover should take place. A mobile switching
centre server is a part of the redesigned MSC concept starting from 3GPP Release 5.
Tasks of the MSC include:

• Delivering calls to subscribers as they arrive based on information from the VLR.
• Connecting outgoing calls to other mobile subscribers or the PSTN.
• Delivering SMSs from subscribers to the short message service centre (SMSC) and vice
versa.
• Arranging handovers from BSC to BSC.
• Carrying out handovers from this MSC to another.
• Supporting supplementary services such as conference calls or call hold.
• Generating billing information

Mobile switching centre server (MSS)

The mobile switching centre server is a soft-switch variant of the mobile switching centre,
which provides circuit-switched calling, mobility management, and GSM services to the mobile
phones roaming within the area that it serves. MSS functionality enables split between control
(signaling) and user plane (bearer in network element called as media gateway/MG), which
guarantees more optimal placement of network elements within the network.
Home location register (HLR)
The home location register (HLR) is a central database that contains details of each mobile
phone subscriber that is authorized to use the network. There can be several logical, and
physical, HLRs per public land mobile network (PLMN), though one international mobile
subscriber identity (IMSI)/MSISDN pair can be associated with only one logical HLR which can
span several physical nodes at a time.
The HLR stores details of every SIM card issued by the mobile phone operator. Each SIM has a
unique identifier called an IMSI which is the primary key to each HLR record.
The next important items of data associated with the SIM are the MSISDNs, which are the
telephone numbers used by mobile phones to make and receive calls. The primary MSISDN is
the number used for making and receiving voice calls and SMS, but it is possible for a SIM to
have other secondary MSISDNs associated with it for fax and data calls. Each MSISDN is also a
primary key to the HLR record. The HLR data is stored for as long as a subscriber remains with
the mobile phone operator.
The HLR is a system which directly receives and processes MAP transactions and messages
from elements in the GSM network, for example, the location update messages received as
mobile phones roam around.
The HLR connects to the following elements:

• The G-MSC for handling incoming calls

• The VLR for handling requests from mobile phones to attach to the network
• The SMSC for handling incoming SMS
• The voice mail system for delivering notifications to the mobile phone that a message is
waiting
• The AUC for authentication and ciphering and exchange of data (triplets)

Authentication centre (AUC)

The authentication centre (AUC) is a function to authenticate each SIM card that attempts to
connect to the network, typically when the phone is powered on. Once the authentication is
successful, the HLR is allowed to manage the SIM and services described above. An encryption
key is also generated that is subsequently used to encrypt all wireless communications (voice,
SMS, etc.) between the mobile phone and the network.
If the authentication fails, then no services are possible from that particular combination of SIM
card and mobile phone operator attempted. Proper implementation of security in and around the
AUC is a key part of an operator's strategy to avoid SIM cloning.
The AUC does not engage directly in the authentication process, but instead generates data
known as triplets for the MSC to use during the procedure. The security of the process depends
upon a shared secret between the AUC and the SIM called the Ki. 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 the IMSI to produce a
challenge/response for identification purposes and an encryption key called Kc for use in over the
air communications.
The AUC connects to the following elements:

• The MSC which requests a new batch of triplet data for an IMSI after the previous data
have been used. This ensures that same keys and challenge responses are not used twice
for a particular mobile.

Visitor location register (VLR)

The visitor location register is a temporary database of the subscribers who have roamed into
the particular area which it serves. Each base station in the network is served by exactly one
VLR; hence a subscriber cannot be present in more than one VLR at a time.
The data stored in the VLR has either been received from the HLR, or collected from the MS. In
practice, for performance reasons, most vendors integrate the VLR directly to the V-MSC and,
where this is not done, the VLR is very tightly linked with the MSC via a proprietary interface.
The VLR connects to the following elements:

• The V-MSC to pass needed data for its procedures; e.g., authentication or call setup.
• The HLR to request data for mobile phones attached to its serving area.
• Other VLRs to transfer temporary data concerning the mobile when they roam into new
VLR areas. For example, the temporal mobile subscriber identity (TMSI).

Equipment identity register (EIR)

The equipment identity register is often integrated to the HLR. The EIR keeps a list of mobile
phones (identified by their IMEI) which are to be banned from the network or monitored. This is
designed to allow tracking of stolen mobile phones. In theory all data about all stolen mobile
phones should be distributed to all EIRs in the world through a Central EIR. The EIR data does
not have to change in real time, which means that this function can be less distributed than the
function of the HLR. The EIR is a database that contains information about the identity of the
mobile equipment that prevents calls from stolen, unauthorized or defective mobile stations.
Some EIR also have the capability to log Handset attempts and store it in a log file.

The Operation and Support System

The operations and maintenance center (OMC) is connected to all the mobile equipments in
the switching system and BSC. This implementation of OMC is operation and support system
(OSS). It is the functional entity by which the network operator is able to monitor and control the
system. The purpose of OSS is to offer the customer cost-effective support for centralized,
regional and local operational and maintenance activities that are required for a GSM network.
The other important function of OSS is to maintain continuous overview of the network and
organizations.

CALL PROCESSING

The processing of a call involves the four states:-

1) Initialization state
2) Mobile Idle state
3) Access state
4) Traffic channel state

In the initialization state, after determining that there is a digital system, the handset
monitors the paging channel, in determining the start and finish of the pilot channel, it can
determine the timing of the sync channel. Once it can read the sync timing, it can further
refine its timing.

During the idle state, the mobile will monitor the paging channel. On the paging channel are
various messages pertaining to set up and operation.

The mobile transmits the access channel at varying slots. The lengths of the slots are
configurable via the access parameter message.

In the traffic channel state we begin transmitting the traffic channel. Traffic channels
contain both voice data as well as the signaling information. There is an extensive listing of
various signaling messages from order messages and service response messages to various
flash and power measurement messages. Upon release of a call, the mobile goes back to the
initialization state.
This is the basic call processing loop.

After power up, the initialization state determines which system to use, whether analog or
CDMA and if it is CDMA it goes into pilot and sync processing.
Once the system is synchronized, the system goes into the mobile station idle state, where it
monitors the paging channel. If a call is to be originated, or the mobile is paged the system
goes into the access state.
Once a call is set up, the phone moves over to the traffic channel state, where the forward and
reverse traffic channels are used to communicate voice and messaging
TRANSMISSION MEDIA
A transmission medium is a material substance which can propagate energy waves. It can also
refer to the technical device which employs the material substance to transmit or guide the
waves. Thus an optical fiber or a copper cable can be referred to as a transmission medium. The
absence of a material medium that is, vacuum can also be thought of as a transmission medium
for electromagnetic waves such as light and radio waves.
For telecommunications purposes, transmission media are broadly classified as:-

• Guided (or bounded) - Waves are guided along a solid medium such as a transmission
line.
• Wireless (or unguided) – Transmission and reception are achieved by means of an
antenna.

Different transmission media which can be used in transmitting information involves:-

Copper wire

In olden days, copper wire was the only means of transmitting information. Technically, known
as unshielded twisted pair (UTP) consisted of a large number of pairs of copper wire of varying
size in a cable. The cable did not have the shield and therefore the signal-primarily the high
frequency part of the signal-was able to leak out. Also, the twisting on the copper pair was very
casual, designed as much to identify which wires belong to a pair as to handle transmission
problems.

Coaxial cable

It is an adaptation of copper wire. Coaxial cable consists of a single strand of copper running
down the axis of the cable. This strand is separated from the outer shielding by an insulator made
of foam or other dielectrics. A conductive shield covers the cable. Because of the construction of
the cable, very high frequencies can be carried without leaking out. In fact, dozens of TV
channels, each 6 MHz wide, can be carried on a single cable.

Microwave

Microwave communication is used in wireless communication where transfer of information

over a distance is without the use of electrical conductors or wires. Microwaves are
electromagnetic waves with wavelengths ranging from 1m down to 1mm or equivalently with
frequencies between 0.3 GHz to 300 GHz. In these frequencies the microwave band can provide
highly reliable, carrier-class, high-capacity communications at ranges of a mile or two for the
high frequencies, to 20 or 30 miles for the lower frequencies.

Within the microwave band, each of the different frequencies has its own channel plan. The
width of the channel in megahertz, along with the modulation used by the radio equipment,
determines the capacity of the link.

The microwave band is used for many applications, such as earth-to-space satellite links and
point-to-point terrestrial communications.

Major components required for microwave communication includes:

• Antenna
• Indoor Unit (IDU)
• Outdoor Unit (ODU)
• Cable normally a coaxial cable

Engineering the microwave link:

An RF engineer begins a design by doing a link budget analysis. A given radio system has a
system gain that depends on the design of the radio and the modulation used. The gains from the
antenna at each end are added to this gain. Larger antennas provide higher gain. The free-space
loss of the radio signal as it travels over the air is then subtracted from the system: the longer the
link the higher the loss. These calculations result in a “fade margin” for the link. Anything that
affects the radio signal within this margin will be overcome by the radio; if the margin is
exceeded, then the link could go down. The next step, then, is to analyze impediments that could
potentially affect the radio signal. With good understanding of the potential effects on the signal,
the RF engineer can design links with availability and performance equal to or better than a wire-
line link.

The major issues with microwave link engineering are rain fade, multipath, and interference.

The energy of microwave radio signals is absorbed by rain; rain can cause an outage if enough
energy is absorbed so that the receiver loses the signal. The engineering of radio links to
accommodate the effects of rain attenuation is based on ITU or Crane global rain models.
These models are derived from meteorological observations combined with the attenuation
estimations for rain rate and frequency. Rain fade is most pronounced at 38 GHz, and has less
of an effect at lower frequencies.
Multipath is a phenomenon that affects the lower frequencies more than the higher frequencies.
A multipath condition occurs when the radio receiver receives multiple signals – the direct signal
plus other signals from the same transmitters that reflect off the terrain or the atmosphere – that
causes the receiver to lose the direct signal. Multipath is combated by careful analysis of the
terrain to place the antennas at a height where the multipath will not be disruptive. Atmospheric
multipath is dynamic and cannot be engineered in this way; in regions where atmospheric
multipath is problematic, two radio links will be deployed in parallel, at varying heights such that
one link is always free of multipath. “Hitless switching” between the links ensures that no data
is lost in multipath conditions.

Interference comes from other radios systems using the same channel or an adjacent channel. A
radio receiver has a certain carrier-to-interference (C/I) ratio that must be maintained. As long
as the RF engineer knows the location and characteristics of potentially interfering links, the new
link can be designed (including channel selection) to always maintain the minimum C/I ratio.
Since most microwave frequencies are licensed by the national regulatory agency, this
information is freely available to the RF engineer, and interference is rarely a problem.

The RF engineer can rely on modern radio systems that have built-in adjustments to overcome
rain fade and interference. Automatic Transmit Power Control (ATPC) keeps the transmitting
power low in clear conditions, which minimizes the interfering effect of the link on other nearby
links. If rain or interference is detected, the radio system will raise it’s transmit power to
overcome the problem. The algorithm is closed-loop in nature, so the transmit power will
continually be adjusted up and down as conditions change.

Microwave is an excellent solution for many communications applications. If engineered properly,

microwave links can provide availability, bit error rate, and reliability that is equivalent to fiber and
suitable for any carrier-class application.

Common applications for microwave include:

• Mobile backhaul – 80% of cellular tower worldwide use microwave links to connect the base
station back to the transmission network. 2xT1/E1 and 4xT1/E1 radios are used at individual
base stations, and DS3/E3 and OC-3/STM-1 radios are used in the transmission backbone
• Metro access – microwave links are used by incumbent and competitive carriers to provide
high-speed access to office buildings
• Long-haul backbone – while fiber has generally replaced microwave on high-capacity long-
haul routes, microwave links are common for backbone links to smaller cities and towns, and
where the terrain is difficult
• Enterprise – Many enterprises, hospitals, and universities use microwave links to connect
campuses, bypassing the telephone company
• Control and Monitoring – Utility companies, railways, and pipeline companies use
microwave to control and monitor their infrastructure.
Fiber

Fiber-optic communications is based on the principle that light in a glass medium can carry
more information over longer distances than electrical signals can carry in a copper or coaxial
medium or radio frequencies through a wireless medium. The purity of today’s glass fiber,
combined with improved system electronics, enables fiber to transmit digitized light signals
hundreds of kilometers without amplification. With few transmission losses, low interference,
and high bandwidth potential, optical fiber is an almost ideal transmission medium.

Principle:-

The operation of an optical fiber is based on the principle of total internal reflection. Light
reflects (bounces back) or refracts (alters its direction while penetrating a different medium),
depending on the angle at which it strikes a surface. Controlling the angle at which the light
waves are transmitted makes it possible to control how efficiently they reach their destination.
Light waves are guided through the core of the optical fiber in much the same way that radio
frequency (RF) signals are guided through coaxial cable. The light waves are guided to the other
end of the fiber by being reflected within the core. The composition of the cladding glass relative
to the core glass determines the fiber’s ability to reflect light. That reflection is usually caused by
creating a higher refractive index in the core of the glass than in the surrounding cladding glass,
creating a “waveguide”.

Design of fiber core and cladding:-

An optical fiber consists of two different types of highly pure, solid glass, composed to form the
core and cladding. A protective acrylate coating then surrounds the cladding. In most cases, the
protective coating is a dual layer composition.
A protective coating is applied to the glass fiber as the final step in the manufacturing process.
This coating protects the glass from dust and scratches that can affect fiber strength. This
protective coating can be comprised of two layers: a soft inner layer that cushions the fiber and
allows the coating to be stripped from the glass mechanically and a harder outer layer that
protects the fiber during handling, particularly the cabling, installation, and termination
processes.

Single mode and multi-mode fibers:-

There are two general categories of optical fiber: single-mode and multimode.

Multimode fiber was the first type of fiber to be commercialized. It has a much larger core than
single-mode fiber, allowing hundreds of modes of light to propagate through the fiber
simultaneously. Additionally, the larger core diameter of multimode fiber facilitates the use of
lower-cost optical transmitters and connectors.

Single-mode fiber, on the other hand, has a much smaller core that allows only one mode of
light at a time to propagate through the core. Single mode fibers have higher capacity. They are
designed to maintain spatial and spectral integrity of each optical signal over longer distances,
allowing more information to be transmitted. Its tremendous information-carrying capacity and
low intrinsic loss have made single-mode fiber the ideal transmission medium for a multitude of
applications.

Single-mode fiber is typically used for longer-distance and higher-bandwidth applications.

Multimode fiber is used primarily in systems with short transmission distances (under 2 km),
such as premises communications, private data networks, and parallel optic applications.

Optical fiber sizes:-

The international standard for outer cladding diameter of most single-mode optical fibers is 125
microns (μm) for the glass and 245 μm for the coating. This standard is important because it
ensures compatibility among connectors, splices, and tools used throughout the industry.
Standard single-mode fibers are manufactured with a small core size, approximately 8 to 10 μm
in diameter. Multimode fibers have core sizes of 50 to 62.5 μm in diameter

Single-mode fiber performance characteristics:-

Optical fiber performance parameters can vary significantly among fibers from different
manufacturers in ways that can affect system’s performance. It is important to understand how to
specify the fiber that best meets system requirements.

Attenuation:-

Attenuation is the reduction of signal strength or light power over the length of the light-carrying
medium. Fiber attenuation is measured in decibels per kilometer (dB/km).
Optical fiber offers superior performance over other transmission media because it combines
high bandwidth with low attenuation. This allows signals to be transmitted over longer distances
while using fewer regenerators or amplifiers, thus reducing cost and improving signal reliability.
Attenuation of an optical signal varies as a function of wavelength.

Attenuation is very low, as compared to other transmission media (i.e., copper, coaxial cable,
etc.), with a typical value of 0.35 dB/km at 1300 nm for standard single-mode fiber. Attenuation
at 1550 nm is even lower, with a typical value of 0.25 dB/km. This gives an optical signal,
transmitted through fiber, the ability to travel more than 100 km without regeneration or
amplification.
Attenuation is caused by several different factors, but primarily scattering and absorption. The
scattering of light from molecular level irregularities in the glass structure leads to the general
shape of the attenuation curve.
Further attenuation is caused by light absorbed by residual materials, such as metals or water
ions, within the fiber core and inner cladding. It is these water ions that cause the “water peak”
region on the attenuation curve, typically around 1383 nm. The removal of water ions is of
particular interest to fiber manufacturers as this “water peak” region has a broadening effect and
contributes to attenuation loss for nearby wavelengths. Some manufacturers now offer low water
peak single-mode fibers, which offer additional bandwidth and flexibility compared with
standard single-mode fibers. Light leakage due to bending, splices, connectors, or other outside
forces are other factors resulting in attenuation.

Dispersion:-

Dispersion is the time distortion of an optical signal that results from the time of flight
differences of different components of that signal, typically resulting in pulse broadening. In
digital transmission, dispersion limits the maximum data rate, the maximum distance, or the
information-carrying capacity of a single-mode fiber link. In analog transmission, dispersion can
cause a waveform to become significantly distorted and can result in unacceptable levels of
composite second-order distortion (CSO).
Single-mode fiber dispersion varies with wavelength and is controlled by fiber design. The
wavelength at which dispersion equals zero is called the zero-dispersion wavelength (λ 0). This
is the wavelength at which fiber has its maximum information-carrying capacity. For standard
single-mode fibers, this is in the region of 1310 nm.

Chromatic dispersion consists of two kinds of dispersion. Material dispersion refers to the
pulse spreading caused by the specific composition of the glass.
Waveguide dispersion results from the light traveling in both the core and the inner cladding
glasses at the same time but at slightly different speeds. The two types can be balanced to
produce a wavelength of zero dispersion anywhere within the 1310 nm to 1650 nm operating
window.

Mode-field diameter:-

Mode-field diameter (MFD) describes the size of the light-carrying portion of the fiber. For
single-mode fibers, this region includes the fiber core as well as a small portion of the
surrounding cladding glass. MFD is an important parameter for determining a fiber’s resistance
to bend-induce loss and can affect splice loss as well. MFD is the functional parameter that
determines optical performance when a fiber is coupled to a light source, connectorized, spliced,
or bent. It is a function of wavelength, core diameter, and the refractive-index difference between
the core and the cladding. These last two are fiber design and manufacturing parameters.

Environmental performance:-

While cable design and construction play a key role in environmental performance, optimum
system performance requires the user to specify fiber that will operate without undue loss from
micro bending. Micro bends are small-scale perturbations along the fiber axis, the amplitude of
which are on the order of microns. These distortions can cause light to leak out of a fiber. Micro
bending may be induced at very cold temperatures because the glass has a different coefficient of
thermal expansion from the coating and cabling materials. At low temperatures, coating and
cables become more rigid and may contract more than the glass. Consequently, enough loads
may be exerted on the glass to cause microbends. Coating and cabling materials are selected by
the manufacturers to minimize loss due to micro bending.

TRANSMISSION TECHNOLOGY:-
Most transmission in the local exchange plant is analog in nature. That is, the signal being
transmitted varies continuously, both in frequency and in amplitude. A high-pitched voice mostly
contains high frequencies; a low-pitched voice, low frequencies. A loud voice has a high
amplitude signal and a soft voice has a low-amplitude signal. In the long-distance network, and
more and more in the local exchange plant, digital transmission is being used. A digital signal is
comprised of a stream of 1s and 0s that portray the analog voice signal by means of a code.
Analog signals can be combined, i.e., multiplexed, by combining them with a “carrier”
frequency.

There are various techniques involved in multiplexing:-

FDMA - Frequency Division Multiple Access

FDMA is used for standard analog cellular. Each user is assigned a discrete slice of the RF
spectrum. FDMA permits only one user per channel since it allows the user to use the channel
100% of the time. Therefore, only the frequency “dimension” is used to define channels. Here,
the channel bandwidth would be 30 KHz for AMPS and 25 KHz for TACS.
TDMA - Time Division Multiple Access

Here, users are still assigned a discrete slice of RF spectrum, but multiple users can now share
that RF carrier on a time slot basis. Each of the users alternates their use of the RF channel.
Frequency division is still employed, but these carriers are now further sub-divided into some
number of time slots per carrier. A user is assigned a particular time slot in a carrier and can only
send or receive information at those times. Information flow is not continuous for any user, but
rather is sent and received in “bursts.” The bursts are re-assembled at the receiving end, and
appear to provide continuous sound because the process is very fast.
GSM

GSM (Global System for Mobile communications) is the most popular standard for mobile
phones in the world. GSM is used by over 3 billion people across more than 212 countries and
territories. Its ubiquity makes international roaming very common between mobile phone
operators, enabling subscribers to use their phones in many parts of the world. GSM differs from
its predecessors in that both signaling and speech channels are digital, and thus is considered a
second generation (2G) mobile phone system. This has also meant that data communication was
easy to build into the system.
GSM pioneered a low-cost alternative to voice calls, the Short message service (SMS, also
called "text messaging"), which is now supported on other mobile standards as well. Also, the
standard includes one worldwide Emergency telephone number, 112. This makes it easier for
international travelers to connect to emergency services without knowing the local emergency
number.
Newer versions of the standard were backward-compatible with the original GSM phones. For
example, Release '97 of the standard added packet data capabilities, by means of General Packet
Radio Service (GPRS). Release '99 introduced higher speed data transmission using Enhanced
Data Rates for GSM.

Cellular radio network

GSM is a cellular network, which means that mobile phones connect to it by searching for cells
in the immediate vicinity.
There are five different cell sizes in a GSM network- macro, micro, Pico, Femto and umbrella
cells. The coverage area of each cell varies according to the implementation environment. Macro
cells can be regarded as cells where the base station antenna is installed on a mast or a building
above average roof top level. Micro cells are cells whose antenna height is under average roof
top level; they are typically used in urban areas. Pico cells are small cells whose coverage
diameter is a few dozen meters; they are mainly used indoors. Femto cells are cells designed for
use in residential or small business environments and connect to the service provider’s network
via a broadband internet connection. Umbrella cells are used to cover shadowed regions of
smaller cells and fill in gaps in coverage between those cells.
Cell horizontal radius varies depending on antenna height, antenna gain and propagation
conditions from a couple of hundred meters to several tens of kilometers. The longest distance
the GSM specification supports in practical use is 35 kilometers.
The modulation used in GSM is Gaussian minimum-shift keying (GMSK), a kind of
continuous-phase frequency shift keying. In GMSK, the signal to be modulated onto the carrier
is first smoothed with a Gaussian low-pass filter prior to being fed to a frequency modulator,
which greatly reduces the interference to neighboring channels (adjacent channel interference).

Interference with audio devices

Some audio devices are susceptible to radio frequency interference (RFI), which could be
mitigated or eliminated by use of additional shielding or bypass capacitors in these audio
devices. However, the increased cost of doing so is difficult for a designer to justify
It is a common occurrence for a nearby GSM handset to induce a kind of disturbance in audio
output on PAs, wireless microphones, home stereo systems, televisions, computers, cordless
phones, and personal music devices. When these audio devices are in the near field of the GSM
handset, the radio signal is strong enough that the solid state amplifiers in the audio chain act as a
detector. The clicking noise itself represents the power bursts that carry the TDMA signal. These
signals have been known to interfere with other electronic devices, such as car stereos and
portable audio players. This also depends on the handset's design, and its conformance to strict
rules and regulations allocated by the US body, the FCC, in part 15 of its rules and regulations
pertaining to interference with electronic devices.
GSM frequencies
GSM networks operate in a number of different frequency ranges (separated into GSM
frequency ranges for 2G and UMTS frequency bands for 3G). Most 2G GSM networks operate
in the 900 MHz or 1800 MHz bands. Some countries in the Americas (including Canada and the
United States) use the 850 MHz and 1900 MHz bands because the 900 and 1800 MHz frequency
bands were already allocated. Most 3G GSM networks in Europe operate in the 2100 MHz
frequency band..
GSM-900 uses 890–915 MHz to send information from the mobile station to the base station
(uplink) and 935–960 MHz for the other direction (downlink), providing 125 RF channels
(channel numbers 0 to 124) spaced at 200 kHz. Duplex spacing of 45 MHz is used.
In some countries the GSM-900 band has been extended to cover a larger frequency range. This
'extended GSM', E-GSM, uses 880–915 MHz (uplink) and 925–960 MHz (downlink), adding 50
channels (channel numbers 975 to 1023 and 0) to the original GSM-900 band. Time division
multiplexing is used to allow eight full-rate or sixteen half-rate speech channels per radio
frequency channel. There are eight radio timeslots (giving eight burst periods) grouped into what
is called a TDMA frame. Half rate channels use alternate frames in the same timeslot. The
channel data rate for all 8 channels is 270.833 kbit/s, and the frame duration is 4.615 ms.
The transmission power in the handset is limited to a maximum of 2 watts in GSM850/900 and 1
watt in GSM1800/1900.

Voice codecs

GSM has used a variety of voice codecs to squeeze 3.1 kHz audio into between 5.6 and 13 kbit/s.
Originally, two codecs, named after the types of data channel they were allocated, were used,
called Half Rate (5.6 Kbit/s) and Full Rate (13 kbit/s). These used a system based upon linear
predictive coding (LPC). In addition to being efficient with bitrates, these codecs also made it
easier to identify more important parts of the audio, allowing the air interface layer to prioritize
and better protect these parts of the signal.
GSM was further enhanced in 1997 with the Enhanced Full Rate (EFR) codec, a 12.2 kbit/s
codec that uses a full rate channel. Finally, with the development of UMTS, EFR was refactored
into a variable-rate codec called AMR-Narrowband, which is high quality and robust against
interference when used on full rate channels, and less robust but still relatively high quality when
used in good radio conditions on half-rate channels.

Subscriber Identity Module (SIM)

One of the key features of GSM is the Subscriber Identity Module, commonly known as a SIM
card. The SIM is a detachable smart card containing the user's subscription information and
phone book. This allows the user to retain his or her information after switching handsets.
Alternatively, the user can also change operators while retaining the handset simply by changing
the SIM. Some operators will block this by allowing the phone to use only a single SIM, or only
a SIM issued by them; this practice is known as SIM locking, and is illegal in some countries.
The locking applies to the handset, identified by its International Mobile Equipment Identity
(IMEI) number, not to the account (which is identified by the SIM card).

GSM security

GSM was designed with a moderate level of security. The system was designed to authenticate
the subscriber using a pre-shared key and challenge-response. Communications between the
subscriber and the base station can be encrypted. The development of UMTS introduces an
optional USIM, that uses a longer authentication key to give greater security, as well as mutually
authenticating the network and the user - whereas GSM only authenticates the user to the
network (and not vice versa). The security model therefore offers confidentiality and
authentication, but limited authorization capabilities, and no non-repudiation.

GSM Subscriber Services:-

There are two basic types of services offered through GSM: telephony and data. Telephony
services are mainly voice services that provide subscribers with the complete capability,
including necessary terminal equipment, to communicate with other subscribers. Data services
provide the capacity necessary to transmit appropriate data signals to access points creating an
interface to the network. In addition, to normal telephony and emergency calling, the following
subscriber services are supported by GSM:-

• DUAL-TONE MULTI FREQUENCY (DTMF):-

 DTMF is a tone signaling scheme often used for various control purposes via the
telephone network, such as remote control of an answering machine. GSM supports full
originating DTMF.
• Facsimile group III:-
GSM supports CCITT group III facsimile. As standard FAX machines are designed to be
connected to a telephone using analog signals, a special FAX converter connected to the
exchange is used in the GSM system. This enables the GSM-connected FAX TO
communicate with any analog FAX in the network.
• Short message services:-
A message consisting of a maximum of 160 alphanumeric characters can be sent to or
from a mobile station. This service can be viewed as an advanced form of alphanumeric
paging with a number of advantages. If the subscriber’s mobile unit is powered off or has
left the coverage area, the message is stored and offered back to the subscriber when the
mobile is powered on or has re-entered the coverage area of the network.
• Cell broadcast:-
A variation of the short message service is the cell broadcast facility. A message of a
maximum of 93 characters can be broadcast to all mobile subscribers in a certain
geographic area. Typical applications include traffic congestion warnings and reports on
accidents.
• Voice mail:-
This service is actually an answering machine within the network, which is controlled by
the subscriber. Calls can be forwarded to the subscriber’s voice mail-box and the
subscriber checks for messages via a personal security code.
• Fax mail:-
With this service, the subscriber can receive fax messages at any fax machine. The
messages are stored in a service center from which they can be retrieved by the
subscriber via a personal security code to the desired fax number.

GSM also supports a comprehensive set of supplementary services that can complement and
support both telephony and data services. Supplementary services are defined by GSM and
characterized as revenue generating features. Some of them are:- call forwarding, barring of
outgoing calls, barring of incoming calls, advice of charge (AOC), call hold, call waiting,
multiparty service, calling line identification presentation/restriction, closed user groups(CUGs)
etc.

Advantages:

• GSM is mature; this maturity means a more stable network with robust features.
• Less signal deterioration inside buildings.
• Ability to use repeaters.
 • Talk-time is generally higher in GSM phones due to the pulse nature of transmission.
• The availability of Subscriber Identity Modules allows users to switch networks and
handsets at will, aside from a subsidy lock.
• GSM covers virtually all parts of the world so international roaming is not a problem.
• The much bigger number of subscribers globally creates a better network effect for GSM
handset makers, carriers and end users.

Disadvantages:-

• Pulse nature of TDMA transmission used in 2G interferes with some electronics,

especially certain audio amplifiers. 3G uses W-CDMA now.
• Intellectual property is concentrated among a few industry participants, creating barriers
to entry for new entrants and limiting competition among phone manufacturers.
• GSM has a fixed maximum cell site range of 35 km, which is imposed by technical
limitations.

CDMA:-

Code Division Multiple Access (CDMA) is a spread-spectrum multiple access technique. This
means a large number of users share a common pool of radio channels with any user being able
to get access to any channel. With CDMA, unique digital codes, rather than separate RF
frequencies or channels, are used to differentiate subscribers. The codes are shared by both the
mobile station (cellular phone) and the base station, and are called “pseudo-Random Code
Sequences.” All users share the same range of radio spectrum.
There are several types of CDMA but the most commonly used variant for cellular is Direct
Sequence CDMA (DS-CDMA). Each user is assigned a binary, Direct Sequence code during a
call. The DS code is a signal generated by linear modulation with wideband Pseudorandom
Noise (PN) sequences. As a result, DS CDMA uses much wider signals than those used in other
technologies. Wideband signals reduce interference and allow one cell frequency reuse.
For cellular telephony, CDMA is a digital multiple access technique specified by the
Telecommunications Industry Association (TIA) as “IS-95”. IS-95 uses a multiple access
spectrum spreading technique called Direct Sequence (DS) CDMA. IS-95 systems divide the
radio spectrum into carriers which are 1,250 kHz (1.25 MHz) wide.

One of the unique aspects of CDMA is that while there are certainly limits to the number of
phone calls that can be handled by a carrier, this is not a fixed number. Rather, the capacity of
the system will be dependent on a number of different factors. Therefore, the maximum number
of users, or effective traffic channels, per carrier depends on the amount of activity that is going
on in each channel, and is therefore not precise. It is a “soft overload” concept where an
additional user (or conversation) can usually be accommodated if necessary, at the “cost” of a bit
more interference to the other users.

Spread spectrum:-

CDMA is a “spread spectrum” technology, which means that it spreads the information
contained in a particular signal of interest over a much greater bandwidth than the original signal.
A CDMA call starts with a standard data rate of 9600 bits/second (9.6 kilobits/second). This is
then spread to a transmitted rate of about 1.23 Megabits/second. Spreading means that digital
codes are applied to the data bits associated with users in a cell. These data bits are transmitted
along with the signals of all the other users in that cell. When the signal is received, the codes are
removed from the desired signal, separating the users and returning the call to a rate of 9600 bps.
Traditional uses of spread spectrum are in military operations. Because of the wide bandwidth of
a spread spectrum signal, it is very difficult to jam, difficult to interfere with, and difficult to
identify. Since a wideband spread spectrum signal is very hard to detect, it appears as nothing
more than a slight rise in the "noise floor" or interference level. With other technologies, the
power of the signal is concentrated in a narrower band, which makes it easier to detect. Increased
privacy is inherent in CDMA technology. CDMA phone calls will be secure from the casual
eavesdropper since, unlike an analog conversation, a simple radio receiver will not be able to
pick individual digital conversations out of the overall RF radiation in a frequency band.

Synchronization

In the final stages of the encoding of the radio link from the base station to the mobile, CDMA
adds a special “pseudo-random code” to the signal that repeats itself after a finite amount of
time. Base stations in the system distinguish themselves from each other by transmitting different
portions of the code at a given time. In other words, the base stations transmit time offset
versions of the same pseudo-random code. In order to assure that the time offsets used remain
unique from each other, CDMA stations must remain synchronized to a common time reference.
The Global Positioning System (GPS) provides this precise common time reference. GPS is a
satellite based radio navigation system capable of providing a practical and affordable means of
determining continuous position, velocity, and time to an unlimited number of users.

Vocoder

A vocoder is a device which can take analog voice, and using various predictive algorithms,
compress and encode this voice data.
Currently, there are 2 vocoders supported in CDMA systems. Originally, the 8k vocoder was to
be the main vocoder. A lower data rate means better system performance.
EVRC is a newly developed enhanced 8k vocoder. It will have improved voice quality than the
current 8k, which can increase system performance.
In a normal voice conversation, one person speaks while one person listens. In, lets say 50% of
the time when it is our turn to speak, our voice patterns only require high speed vocoding in a
portion of our spoken words. Variable vocoders take advantage of this fact by varying the data
rate.

Protocol architecture
Layer 1 deal with the actual radio transmission, frequency use, etc. Layer 2 offers a best effort
delivery of voice and data packets. The MAC sub layer of this layer also performs channel
management. Data originating from different sources are multiplexed and handed for
transmission to the physical layer.

Soft Handoffs

With traditional hard hand-offs, which are used in all other types of cellular systems, the mobile
drops a channel before picking up the next channel. When a call is in a soft hand-off condition, a
mobile user is monitored by two or more cell sites and the transcoder circuitry compares the
quality of the frames from the two receive cell sites on a frame-by-frame basis. The system can
take advantage of the moment-by-moment changes in signal strength at each of the two cells to
pick out the best signal. This ensures that the best possible frame is used in the CDMA decoding
process. The transcoder can literally toggle back and forth between the cell sites involved in a
soft hand-off on a frame-by-frame basis, if that is what is required to select the best frame
possible.
Soft hand-offs also contribute to high call quality by providing a “make before break”
connection. This eliminates the short disruption of speech one hears with non-CDMA
technologies when the RF connection breaks from one cell to establish the call at the destination
cell during a hand-off. In CDMA the cells “team up” to obtain the best possible combined
information stream. CDMA hand-offs do not create the "hole" in speech that is heard in other
technologies.

CDMA Power Control

Open Loop Power Control

Open Loop Power Control is based on the similarity of the loss in the forward path to the loss in
the reverse path. It sets the sum of transmit power and receive power to a constant (- 73 dBm). A
reduction in signal level at the receive antenna will result in an increase in signal power from the
transmitter.

Close Loop Power Control

Close loop power control is used to allow the power from the unit to deviate from the nominal as
set by the open loop control. This is done with a form of delta modulation. The base station
monitors the power received from each mobile station and commands the mobile to either raise
power or lower power by a fixed step of 1 dB. This process is repeated 800 times per second , or
every 1.2ms .
Because the mobile’s power in controlled to be no more than is needed to maintain the link at the
base station , a CDMA mobile typically transmit much less power than an analog phone. The
base station monitors the received quality 800 times per second and directs the mobile to raise or
lower its power until the received signal quality is just adequate.
Most of the time, an analog phone transmit excess power to maintain the link.
.
CDMA Channels

CDMA traffic channels are dependent on the equipment platform, such as Motorola’s SC™
products, on which the CDMA is implemented. Motorola designates channels in three ways:
effective traffic channels, actual traffic channels and physical traffic channels.
• The number of “Effective” traffic channels includes the traffic carrying channels less the
soft hand-off channels. The capacity of an effective traffic channel is equivalent to the
traffic carrying capacity of an analog traffic channel.
• The number of “Actual” traffic channels includes the effective traffic channels plus
channels allocated for soft hand-off.
• The number of “Physical” traffic channels includes the Pilot channels, the Sync channels,
the Paging channels, the Soft Hand-off Overhead channels and the Effective (voice and
data) traffic channels.
• CDMA uses the terms “forward” and “reverse” channels just like they are used in
analog systems. Base transmit equates to the forward direction, and base receive is the
reverse direction. “Forward” is what the subscriber hears and “reverse” is what the
subscriber speaks.

CDMA Forward Channels

Pilot Channel
The pilot channel is used by the mobile station to obtain initial system synchronization and to
provide time, frequency, and phase tracking of signals from the cell site.

Sync Channel
This channel provides cell site identification, pilot transmit power, and the cell site pilot pseudo-
random (PN) phase offset information. With this information the mobile units can establish the
System Time as well as the proper transmits power level to use to initiate a call.

Paging Channel
Once the mobile has obtained the system information that it needs from the sync channel, the
subscriber unit adjusts its timing to correspond to the System Time and begins monitoring the
paging channel. Once a mobile has been paged and acknowledges that page, call setup and traffic
channel assignment information is then passed on this channel to the mobile.

Forward Traffic Channel

This channel carries the actual phone call and carries the voice and mobile power control
information from the base station to the mobile unit.
CDMA Reverse Channels

Access Channel
This channel provides communication from the mobile station to the base station when the
mobile station is not using a traffic channel. The access channel is used for call originations, and
responses to pages, orders, and registration requests. The access channels are paired with a
corresponding paging channel.

Reverse Traffic Channel

This channel carries the other half of the actual phone call and carries the voice and mobile
power control information from the mobile unit to the base station.

Modulation

Both the Forward and Reverse Traffic Channels use a similar control structure consisting of 20
millisecond frames. For the system, frames can be sent at either 14400, 9600, 7200, 4800, 3600,
2400, 1800, or 1200 bps. The receiver detects the rate of the frame and processes it at the correct
rate. This technique allows the channel rate to dynamically adapt to the speech or data activity.
For speech, when a talker pauses, the transmission rate is reduced to a low rate. When the talker
speaks, the system instantaneously shifts to using a higher transmission rate. This technique
decreases the interference to other CDMA signals and thus allows an increase in system
capacity. CDMA starts with a basic data rate of 9600 bits per second. This is then spread to a
transmitted bit rate, or chip rate of 1.2288 MHz The spreading process applies digital codes to
the data bits, which increases the data rate while adding redundancy to the system. The chips are
transmitted using a form of QPSK (quadrature phase shift keying) modulation which has
been filtered to limit the bandwidth of the signal. This is added to the signal of all the other users
in that cell. When the signal is received, the coding is removed from the desired signal, returning
it to a rate of 9600 bps. The ratio of transmitted bits or chips to data bits is the coding gain. The
coding gain for the IS-95 CDMA system is 128, or 21 dB.

Advantages

When implemented in a cellular telephone system, CDMA technology offers numerous benefits
to the cellular operator and their subscribers:-

1) Capacity increases: 8 to 10 times that of an AMPS analog system, and 4 to 5 times that
of a GSM system.

2) Improved call quality: CDMA will provide better and more consistent sound as compared
to AMPS. Cellular telephone systems using CDMA should be able to provide higher
quality sound and phone calls than systems based on other technologies.
 3) Simplified system planning: Engineers will no longer have to perform the detailed
frequency planning which is necessary in analog and TDMA systems.

4) Enhanced privacy: Increased privacy over other cellular systems, both analog and digital,
is inherent in CDMA technology.

5) Increased talk time and standby time for portables: Because of precise power control and
other system characteristics, CDMA subscriber units normally transmit at only a fraction
of the power of analog and TDMA phones.

6) Variable bandwidth offering ease of implementation: CDMA's inherent multipath

resistance and high bandwidth makes it suitable for high bit rate applications.

7) Lower power transmitters:-This results in lower hazard and lower costs. It also reduces
interference with other systems which is a key factor for operators.

8) Improved quality on handoffs: Through the soft handoff process, reliability is improved
for handoffs.

9) Improved coverage characteristics: A CDMA cell site has a greater range than a typical
analog or digital cell site. Therefore fewer CDMA cell sites are required to cover the
same area. Depending on system loading and interference, the reduction in cells could be
as much as 50% when compared to GSM

10) CDMA is one of the systems currently being considered for the Third Generation Mobile
Standard.

Disadvantages

1) CDMA systems require more complexity in both the Base Stations and the Mobile
handsets to handle the unique power control requirements of CDMA and the complicated
encoding/decoding mechanisms.

2) The near-far effect is more significant in CDMA systems. This requires complex
open/closed loop control mechanisms to eliminate the effects.

3) "Rogue Mobiles" i.e. mobiles that don't obey power control commands from the base-
station can cause havoc in CDMA systems. Methods such as malfunction timers and lock
orders need to be introduced.
 4) Soft handoffs, though improving handoff quality, increases the infrastructure cost,
network complexity and maintenance effort. Additional lines between the BTS and BSC
plus additional transceiver circuits are required.

5) In situations of heavy cell loading an effect termed the "breathing effect" is being noticed
whereby the cell effectively contracts. This requires careful positioning of antennae etc.

Difference between GSM and CDMA

Feature GSM CDMA

Technology Digital Circuit Switched technology Digital Circuit Switched

based on TDMA technology based on Spread
Spectrum
Data Rate 9.6 ~ 14.4 kbps 9.6 ~ 14.4 kbps (IS-95A) , 115
kbps (IS-95B)
Modulation GMSK QPSK , OQPSK
Frequency and Carrier 900 MHz or 1.9 GHz 800 MHz or 1.9 GHz
200 KHz 1.25 MHz
Voice Quality Good Good
Coverage 35 km (max) 100 km (max)
System Capacity Good Excellent
Hand-off Hard Soft
Battery Life of handset Excellent Good

International roaming Excellent Limited

Breathing No Yes

Common Interference Interferes with some electronics, None

 such as amplifiers

Operator Locking Un-lockable ESN

Handset Selection Wide choice Limited choice
Major Countries Europe, India USA

World- wide Market 72% 12%

Share

Cellular Subscribers 75% (1.36 Billion) 14% (250 Million)

CDMA REFERENCE ARCHITECTURE

Radio Access Network (RAN)

The Radio Access Network (RAN) provides the basic transmission, local control, and
management functions associated with processing Subscriber Device services. This involves the
management and control of establishing the Subscriber on the Radio Channel, as well as
establishing procedures with the Core and Packet Data Network for network level processing
(e.g., Call Delivery). The RAN also includes the OMC–R. For the purposes of management, the
RAN is broken down into multiple management domains, each comprised of the components of
one or more MTSOs: the O&M controller devices such as the OMCR and CEM, the gateway
devices used for communication with other operators networks or vendors equipment, and the
Transport Network devices that connect all of these end nodes.

Core Network (Core NW)

The Core Network element provides the functional responsibility for establishing connections
into the PSTN for Circuit Oriented Services. The Core NW, more specifically the MSC, manages
Subscriber information (Static and Dynamic). The Core NW is also responsible for the
generation of billing records. The Core NW provides authentication and MS location services via
interactions with HLR and VLR core components.
Packet Network (Packet Data Network)

The Packet Network element provides the functional responsibility for establishing sessions
into the PSTN (e.g., VOIP) or IP Networks for Packet Data Oriented Services. The Packet Data
Network provides basic non–real time data services with the proposed evolution to real–time
based applications, such as VOIP. The Packet Data Network currently relies on the Core NW for
subscriber authentication and location. Authentication of packet data sessions is performed by
the Packet Data Network (AAA server). From a RAN perspective, the Packet NW is viewed as
either Packet IWU based or PDSN based.

Mobile Subscriber (MS)

The Mobile Subscriber (MS) element defines the end point for services. The MS is responsible
for the initiation services. It also processes service requests from the network to establish
services initiated by other end devices. The MS operates a set of procedures to enable the
management and control of services with the system, such as Registration.

Circuit Based RANs

The Circuit Based RAN refers to a remote Motorola Circuit Based Radio Access Network
which supports connections to the RAN for soft handoff support. This Circuit Based RAN refers
to the selected interconnect used between the two CBSCs for Soft Handoff support. For the
Remote CBSC to be a Packet CBSC, the operator must choose a Circuit Based Interconnect for
soft handoff.
Packet Backhaul Based RANs
The Packet Based RAN refers to a remote Motorola Packet Based Radio Access Network,
which supports and connects to the RAN for soft handoff support.

Inter–Vendor RANs
The Inter–Vendor RAN refers to a remote non–Motorola Radio Access Network which
supports connects to the RAN for soft handoff support.

Interfaces
The following provides an overview of the interfaces to the Motorola RAN.

RAN to Core/MSC:

–A1
–A2
–A5
–A1 over TCP/IP

RAN to PDN:

–L–interface to Packet IWU (viewed only for system configurations containing installed Packet
IWUs)
–L–Interface to Circuit IWU
–A10, A11 (R–P) to PDSN

RAN to Mobile Station (MS):

–IS–95 A/B (including HSPD)

–IS–2000

RAN to “circuit based” RAN:

–Anchor PCF support via the Transport Network (ANs)

–Inter–CBSC SHO support via the Transport Network (ANs)
MOTOROLA RADIO–PACKET ACCESS NETWORK
ARCHITECTURE

The various components include:

Access Node (AN)

The Access Node (AN), a set of IP switches and routers, interconnects all of the cellular
infrastructure devices and transports IP–based traffic to and from the CDMA 2000 cellular
network. The AN hardware architecture consists of two custom frames, the Aggregation Point
frame and the IP Switch frame. The Aggregation Point frame houses one or two Aggregation
Points. The IP Switch frame houses two IP Switches. The Aggregation Point provides WAN
connections to the BTSs and supports JT1, T1, E1 and T3. The IP Switch interconnects the other
cellular infrastructure devices and supports various types of Ethernet LAN interfaces. The
Aggregation Point and IP Switch frames, in addition to housing the Cisco equipment, provide
power distribution functions.
Access Node Functions

The functions of the Access Node includes:-

Provide the Transport Network for the CDMA system

The main function of the AN is to provide the Transport Network for the CDMA system. The
Access Node, which consists of a set of Layer 2 switches and Layer 3 routers, performs the
following Transport Network functionality:

• Interconnection all of the CDMA system devices into a hub–like formation, using span
line and Ethernet interfaces.

• Implement BTS backhaul functions by aggregating span links to/from the BTSs.

• The IP Switch performs the Layer 2 switching and Layer 3 routing of all call traffic
(voice, circuit, and packet data calls), control signaling, and O&M needed to support call
processing in the system.

• The AN functions as a pass–through network that interfaces the cellular network to IP

data networks to transport packet data call traffic. The IP data network elements that must
interface to the cellular network are the PDSN, AAA and optionally the HA. The AN can
connect to other networks using LAN and WAN interfaces with Layer 3 routing
functions.

Integrate separate ANs to increase the borders of the system

To integrate separate ANs (local and/or remote), it is necessary to use LAN–WAN routers. The
AN uses the protocols necessary to establish inter–AN communication. A main purpose of
integrating multiple ANs, and therefore multiple BSSANs, under one OMC–R is to extend the
O&M and soft handoff boundaries further across multiple AN areas.

Implement the Switching and Routing Protocols

OSPF (Open Shortest Path First) is the AN’s primary routing protocol. The router, when
receiving data to send to a destination, determines the best/most efficient path to the network
destination based on routing tables that it continually updates with the status of links between
network routers. OSPF requires routers in the network to send messages indicating current status
of links The OSPF protocol supports authentication.
The AN uses other common protocols such as: SNMP (Simple Network Management Protocol)
which is enabled on the system to manage AN performance and troubleshoot network problems,
PPP (Point–to–Point) is a protocol for transporting data across Point–to–Point links through a
WAN to remote networks, TCP, UDP, IGMP, PIM, SCTP, BGP, PPP, HSRP, VTP.

Packet Data Service Node (PDSN):-

The PDSN acts as the gateway between the landline IP PDN and the cellular system. That is, the
PDSN transfers packet data traffic between the cellular system and the landline PDN (Packet
Data Network).
Operators can choose to group from 2 to 12 PDSNs on the same sub–network, forming a PDSN
cluster. Each PDSN in a PDSN cluster periodically broadcasts information about its status and
load to all the other PDSNs in the cluster. These broadcasts enable the cluster–based PDSNs
to:

• Minimize the number of handoffs between PDSNs in the cluster

• Balance subscriber call load across the PDSNs in the cluster

• Dynamically add and remove PDSNs from the cluster to provide for redundancy and
increased service availability

The PDSN, in combination with HA and AAA devices, provides the following types of
Internet/Intranet access services to users accessing the system:

Mobile IP

– An access method allowing a Mobile IP mobile subscriber to access the Internet or an Intranet
through its connection to the PDSN.
– Mobile IP is an access method allowing a mobile to have a static or dynamic IP address
belonging to its home network where it can access the Internet/Intranet and receive data, and also
roam to foreign networks where it can have data packets forwarded from the home address to the
current address.

Simple IP
– An access method allowing a Simple IP mobile subscriber to access the Internet or an Intranet
through its connection to the PDSN.
– The PDSN (or AAA) assigns a temporary IP address to the mobile.
– When the call is finished the PDSN releases the IP address. If the mobile moves out of the
PDSN serving area, the IP address is released and the mobile can no longer receive packets sent
to that address. The mobile must set up a session with the new PDSN.
– The user profile provided by the AAA can contain information for the PDSN to either grant
access to the IP network or set up a tunnel to an ISP or a Private network.

Proxy Mobile IP

An access method allowing a Simple IP mobile subscriber to access the Internet or an Intranet
through its connection to the PDSN, but the PDSN also registers the mobile with the local HA in
order to provide Mobile IP type services.

Selector Distributor Unit (SDU)

The SDU performs the Selection/Distribution function (SDF) and Packet Control function (PCF)
for CDMA voice and packet data calls, and IS–95A/B voice and data calls.
The Selection Distribution Unit or SDU is a new hardware realization of existing function that
has many advantages:

• Increased capacity of CBSC when used with Packet Backhaul

• Client/server model instead of embedded architecture

• Increased XC efficiency by moving functionality to SDU

• No obsolete equipment

• Network level redundancy

• De–couple capacity planning

• Increased network design flexibility

• Simplified Configuration data management

Selection

In a CDMA system, the signal from the handset may be received by multiple BTSs. These
signals are sent to the SDU for selection and the SDU picks the signal that is most usable, using
a complex algorithm that considers multiple factors including signal strength, error rate, and
other factors. The “Selected” signal is then used in the remainder of the system. The selection
process is done every 20 milliseconds to ensure continued signal quality. The information used
for selection is also used for Soft Handoff (SHO).

Vocoder Processing Unit (VPU)

The Vocoder Processing Unit or VPU is a new hardware realization of existing Transcoding
functions.
The VPU has many advantages:

• High Capacity (5000 Erlangs per frame)

• Move from an embedded functionality to a Client – Server model

• High speed Optical interface to MSC

• No obsolete equipment

• Network level redundancya

• Greatly reduced foot print for functionality

• Increased network design flexibility

• Shares some common hardware with SDU to reduce spares expense

• The VPU uses a common platform with significant architectural advantages that adds
additional value in the continuing product evolution.

Functions

The VPU has multiple functions but its name gives its most important function of vocoder
processing. PCM to any selected CDMA vocoding format can be accomplished. The VPU has
fully functional transparency to the existing Transcoder, including echo cancellation, but it uses a
packet interface for compressed mobile traffic rather than circuit. A list of VPU functions is
given below:

• The conversion of vocoded frames to PCM for uplink frames which entail speech
decoding.

• The conversion of PCM to downlink vocoded frames prior to distribution by the SDU
(which entails speech compression) in addition to echo cancellation from the network
side.
 • The support for CDMA 13 Kbps Qualcomm Code Excited Linear Predictive (QCELP)
vocoder, the 8 Kbps Enhanced Variable Rate Coder (EVRC) and the 8 Kbps Basic
Variable Rate vocoder.

• TTY/TDD encoding (forward link) and decoding (reverse link).

• The termination of SONET/SDH OC–3 circuit PCM interfaces from the MSC/DCS.

• The termination of IP over Gigabit Ethernet connection from the Access Node (AN).

• The support of DTMF tone generation in PCM format.

• The support of circuit data calls when the circuit IWU is located at the XC (packet to
circuit interworking only, with no transcoding).

• The support of circuit data calls when the circuit IWU is located at the MSC and the
CBSC contains an XC (ISLP framing over the A5 interface with no transcoding).

• The support of a VPU to SDU interface and a VPU to XC interface.

Operation and Maintenance Centre-Radio (OMC-R)

The OMC–R is a highly available, UNIX–based O&M platform that supports the core
components of the CDMA RAN, including the Central Base Site Controller (CBSC), the Base
Transceiver Stations (BTS), and IP components for circuit and packet networks. This Element
Manager (EM) platform interfaces directly to the elements via Ethernet and acts primarily as a
data collection and mediation device for alarms, events, statistics, and configuration.
The OMC–R consists of a Sun Microsystems E4500 Enterprise server, D1000 Disk Array, and
other third party equipment integrated into a NEBS Level 3 seismic–compliant cabinet. The
OMC–R provides O&M functions for both IS95A/B and CDMA2000 1X network topologies.
Together the OMC–R, OMC–IP, UNO, and SMAP platforms provide total O&M functions for
both the traditional radio network and the new IP network.

Functions:

The OMC–R provides the following primary functions:

• Event/Alarm Management –

OMC–R collects and logs alarm and event information from the RAN and AN devices,
making this information available through ASCII and Common Management Information
 Protocol (CMIP) interfaces. The data can be viewed using tabular or graphical reports on
OMC–R and/or UNO.

• Performance Management – OMC–R provides a temporary store of performance

management (PM) and Call Detail Log (CDL) data that can be uploaded to external
platforms, such as Motorola’s UNO, for longer–term storage and analysis. The OMC–R
can gather other CBSC performance–related information such as CPU utilization, disk
usage, and processor status, creating another avenue for system analysis.

• Security Management – By allowing each operator to be assigned a user ID and

password, OMC–R helps control access to the operations and maintenance functions.
OMC–R’s security management also provides special features for the management of
dial–up lines, and for the management of terminals connected to OMC–R using only on–
premises cabling.

• Configuration Management –

OMC–R has a critical new function: to coordinate configuration management of the

Radio/IP network intersection. That is because each new IP–based RAN deployment or
re–deployment involves configuration on both the radio side and on the IP side. The
OMC–R’s specially designed management tools greatly simplify these deployment
activities, insulating the operators as much as possible from the specifics of the
configuration needed to make the IP RAN elements communicate and operate together.

• Fault Management

This function provides the capability to query and change device states of elements under
OMC–R’s control. It controls RF diagnostics by enabling the execution of loopback and
forward/reverse power tests, allowing operators to selectively test the call–processing
capability of the RAN.

CONCLUSION
Wireless networks constitute an important part of the telecommunications market. The result of
the integration of Internet with mobile system, the wireless Internet, is expected to significantly
increase the demand for wireless data services. The use of wireless transmission and the mobility
of most wireless systems give rise to a number challenges that must be addressed in order to
develop efficient wireless systems. The challenges include wireless medium unreliability,
spectrum use, power management, security, and location or routing. Digital cellular standards
GSM and CDMA meet the current requirements in voice communications and being upgraded to
meet the future demands in mobile multimedia applications. 3G mobile networks represent an
evolution in terms of capacity, data speeds and new service capabilities from second generation
mobile networks to provide an integrated solution for mobile voice and data with wide area
coverage.