What is Handover

HANDOVER :-

In a mobile communications network, the subscriber can move around freely, & to maintain the constant connection with subscriber, so that he can use all his services without any disturbance is done with the help of Hand-Over.

The basic concept is simple - when the subscriber moves from the coverage area of one cell to another, a new connection with the target cell has to be set up and the connection with the old cell has to be released.

There are two reasons for performing a handover: 

1. Handover due to measurements - It occurs when the quality or the strength of the radio signal falls below certain parameters specified in the BSC. The deterioration of the signal is detected by the constant signal measurements carried out by both the mobile station and the BTS. As a consequence, the connection is handed over to a cell with a stronger signal.
2. Handover due to traffic reasons - It occurs when the traffic capacity of a cell has reached its maximum or is approaching it. In such a case, the mobile stations near the edges of the cell may be handed over to neighbouring cells with less traffic load.

The decision to perform a handover is always made by the BSC that is currently serving the subscriber, except for the handover for traffic reasons. In the latter case the MSC makes the decision.

There are four different types of handover

• Intra cell - Intra BSC handover - The smallest of the handovers is the intra cell handover where the Subscriber is handed over to another traffic channel (generally in another frequency) within the same cell. In this case the BSC controlling the cell makes the decision to perform handover.

• 
  
  Inter cell - Intra BSC handover - The subscriber moves from cell 1 to cell 2. In this case the handover process is controlled by BSC. The traffic connection with cell 1 is released when the connection with cell 2 is set up successfully.
  
  

 

• 
  
  Inter cell - Inter BSC handover - The subscriber moves from cell 2 to cell 3, which is served by another BSC. In this case the handover process is carried out by the MSC, but, the decision to make the handover is still done by the first BSC. The connection with the first BSC (and BTS) is released when the connection with the new BSC (and BTS) is set up successfully.
  
  

• Inter MSC handover - The subscriber moves from a cell controlled by one MSC/VLR to a cell in the domain of another MSC/VLR. This case is a bit more complicated. Considering that the first MSC/VLR is connected to the GMSC via a link that passes through PSTN lines, it is evident that the second MSC/VLR can not take over the first one just like that. The MSC/VLR currently serving the subscriber (also known as the anchor MSC), contacts the target MSC/VLR and the traffic connection is transferred to the target MSC/VLR. As both MSCs are part of the same network, the connection is established smoothly. It is important to notice, however, that the target MSC and the source MSC are two telephone exchanges. The call can be transferred between two exchanges only if there is a telephone number identifying the target MSC. Such a situation makes it necessary to generate a new number, the Handover Number (HON).
  

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Common Terms in Signaling Network

• Signaling Point (SP) - The switching or processing node in a signaling network, where the function of SS7 are implemented. like MSC in GSM or Transmit Exchange in PSTN. Every SP is identified by a number determined by Network Identifier (NI) & Signaling Point Code (SPC). 
• Network Identifier (NI) - It provides discrimination between International & National messages or between 2 national signaling switches. SPC uniquely idetifies a SP within the signaling network.
• Originating Point (OP) - The SP at which the signaling message is generated. It is identified by Originating Point Code (OPC).
• Destination Point (DP) - The SP to which the signaling message is destined. It is identified by Destination Point Code (DPC).
• Signaling Transfer Point (STP) - This is a SP, that is able to route signaling messages. In GSM, every SP is STP as soon as it routes the signaling messages that must be delivered to different destination point, in this case, only MTP is used, upper layers are not involved.
• Signaling Link (SL) - The packet data link that connects 2 SPs is a signaling link. It is not neccessary to have a SL in each PCM line, accoding to maximum load, there may be more than one SL between 2 SPs depending on network structure.
• Link Set (LS) - A number of parallel SL connecting the same SP is Signaling LinkSet.
• Signaling Route (SR) - The predetermined path a message takes through the signaling network between OP & DP is called SR. A linkset may carry several SR and hence convey traffic to several destination.
• Signaling Route Set (SRS) - The signaling network groups all SR that may be used for message transferring between OP & DP is SRS. 
• Circuit Intetity Code (CIC) - The message requires for call setup carries an identity. The identity carries a label with source & destination of the message & also the identity of speech trunk the message is refering to, this identity is termed as CIC.

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Job Consultants

Consultants Address :-

NOTE :- Some of the email addresses may not work but most of them still works (from Beginning) & i hope it will help you to find better opportunities.


SORRY Friends, Those 400-500 email addresses of consultants are making this blog slow, so i've removed those address but
Don't you worry, those who require that e-mail addresses just leave a comment with your e-mail ID, i'll mail you those consultants email addresses soon.

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What is H.248 / MEGACO

Media Gateway Control Protocol (Megaco/H.248)

It is one of the signaling protocol like SS7, SIP, etc.

Megaco(H.248) defines the protocol for Media Gateway Controllers to control Media Gateways for the support of multimedia streams across computer networks. It is typically used to provide Voice over Internet Protocol (VoIP) services (voice and fax) between IP networks and the PSTN, or entirely within IP networks.

Media Gateway :-
A media gateway is any device, such as a circuit switch, IP gateway, or channel bank that converts data from the format required for one type of network to the format required for another.

Megaco/H.248 were introduced to inter-network IP and traditional telephony systems and to provide support for large-scale end-to-end deployments. Hence, It enables traditional telephone networks to transmit voice traffic over IP. While other multimedia over IP protocols, such as Session Initiation Protocol (SIP) and H.323, are based on a peer-to-peer architecture,

It specify a master/slave architecture for decomposed gateways. In the master/slave architecture, MGC is the master server and MGs are the slave clients that behave as simple switches.

One MGC can serve multiple MGs.

MGC or “softswitch” is the foundation for next-generation networks offering the intelligence and reliability of the circuit-switched network with the speed and economy of the packet-switched network. MG is the gateway that allows communication between two different networks, e.g., IP and public switched telephone network.

MGs can communicate via a real-time transport protocol (RTP) that provides end-to-end transport functions suitable for applications transmitting real-time data such as interactive audio and video.
RTP service is further augmented by real-time control protocol (RTCP) to allow monitoring the data delivery.


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What is SIGTRAN

Sigtran:
Defination:Sigtran is a standarised way to carry SS7 signal over an IP Back bone.
Benifit:More than 16 sinaling link is used between two point code.

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What is SIP

SIP (Session Initiation Protocol) :-

It is one of the signaling protocol as SS7 & H.248,

It is an application layer protocol that can extablish, modify & terminate sessions or calls. These sessions include multimedia conference, internet telephony, & similiar applications.

SIP is one of the key protocol that implements voice-over IP (VOIP).

It is a signalling protocol used for establishing sessions in an IP network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. The ability to establish these sessions means that a host of innovative services become possible, such as voice-enriched e-commerce, web page click-to-dial, Instant Messaging with buddy lists, and IP Centrex services.

SIP supported services -

• Name Mapping
• Redirection
• ISDN Services
• Intelligent Network (IN) services.
• User location
• User capabilities
• User availability
• Call set-up
• Call handling
• Call forwarding
• Call-forwarding no answer
• Call-forwarding busy
• Call-forwarding unconditional
• Other address-translation services
• Callee and calling "number" delivery, where numbers can be any (preferably unique) naming scheme
• Personal mobility, i.e., the ability to reach a called party under a single, location-independent address even when the user changes terminals
• Terminal-type negotiation and selection: a caller can be given a choice how to reach the party, e.g. via Internet telephony, mobile phone, an answering service, etc.
• Terminal capability negotiation
• Caller and callee authentication
• Blind and supervised call transfer
• Invitations to multicast conference

It support 5 facets of extablishing & terminating multimedia communications :-

1. User Location - Detemining end system to be used for communication.
2. User Capabilities - Determining the media & media parameters tobe used.
3. User Availability - Determining the willingness of called party to engange in communication.
4. Call Setup - Sending ringback tone to the called party & estabilishing call parameters at both end called & calling party.   
5. Call Holding & Control - Includes redirection, transfer & termination of calls.

SIP can also initiate multi-party calls using multipoint control unit (MCU) or fully meshed interconnection.
Internet Telephony gateway that connects PSTN parties can also use SIP to setup calls between them.
SIP can use User Datagram Protocol (UDP) & Transmision Control Protocol (TCP) as transport protocol, UDP is preferred.

Architecture
There are two basic components within SIP:

• SIP user agent.
• SIP network server.

The User Agent is the end system component for the call. The user agent itself has a client element, the User Agent Client (UAC) and a server element, the User Agent Server (UAS). The client element initiates the calls and the server element answers the calls. This allows peer-to-peer calls to be made using a client-server protocol.

SIP user agents can be lightweight clients suitable for embedding in end-user devices such as mobile handsets or PDAs. Alternatively, they can be desktop applications that bind with other software applications such as contact managers.


The SIP server is the network device that handles the signalling associated with multiple calls. The main function of the SIP servers is to provide name resolution and user location, since the caller is unlikely to know the IP address or host name of the called party, and to pass on messages to other servers using next hop routing protocols.

SIP servers can operate in two different modes: stateful and stateless. The difference between these modes is that a server in a stateful mode remembers the incoming requests it receives, along with the responses it sends back and the outgoing requests it sends on.

A server acting in a stateless mode forgets all information once it has sent a request. These stateless servers are likely to be the backbone of the SIP infrastructure while stateful-mode servers are likely to be the local devices close to the user agents, controlling domains of users.

 
SIP Addressing :-
Uniform Resource Locator (URL) are used within SIP messages to indicate the originator (FROM), current destination (requested URL), final destination (TO) of a SIP request & to specify redirection address (Contact).

SIP URL has a Syntax :-

SIP:User:password@host:port;transport-param|user-param|method-param|ttl-param|maddr-param|other-param

Their meaning -

• SIP - indicates SIP is used for communication with a specified end system.
• User - Consists of any characters in the form of email address or telephone number.
• Password - can be included but not recommended because of security risk.
• Host - can be host(other user) domain name or IP address.
• Port - indicates port number to which request is sent, default is 5060, a public SIP port number.
• Transport-Param - Indicates which transport protocol to be used, TCP or UDP, default is UDP.
• User-Param - can be a telephone number, 2 values are available for this field, IP & Phone number, when field is set to "phone" username is telephone number & corresponding end system is an IP Telephony Gateway. 
• Method-Param - Specifies method or operation to be used.
• TTL-Param - Designates the Time-To-Live (TTL) of UDP multicast data packet. It is valid only when transport parameter is UDP & Maddr parameter is "Multicast Address".
• Maddr-Param - Provides the server address to be contacted for a user, overriding the address supplied in the host field. This address is typically a multicast address.

NOTE - The following parameters are optional
Transport-Param, User-Param, Method-Param, TTL-Param, Maddr-Param, Other-Param.


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What is SS7

SS7:- It is a Common Channel Signaling system designed by ITU –T in response to a demand for more features & integrated data services. It defines the architecture, procedures & protocols for information exchange over digital network. It is designed to support call setup, routing, billing, database information, & special services of PSTNs.

The gateway office uses application protocol of SS7 for more than one interface, such as MAP for C/D, BSAP for A interface.

It is classified into 2 parts

• User Part
• MTP (Message Transfer Part)

MTP: - It is responsible for transmitting signaling messages for its users. It ensures reliable signaling message transfer over signaling network by avoiding or minimizing message loss, duplicate or out-of-sequence in case of any system fault or signaling network fault.

It consists of 3 functional layers:-
1. Signaling data link function (MTP-1)
2. Signaling link function (MTP-2)
3. Signaling network function (MTP-3)

MTP-1:- It defines the physical, electrical & functional characteristics of a signaling data link, as well as access method. It is same as physical layer of OSI model. It used to generate & receive signals on physical channels.

MTP-2:- It corporate with MTP-1 to provide a signaling link for reliable signaling message transfer between 2 signaling points. Its function includes signal unit delimitation, signal unit alignment, error detection & correction, initial alignment, flow control & signaling link error monitoring.

MTP-3:- It enables management message transmission between signaling point for the purpose of ensuring a reliable transfer of signaling message over the signaling network, in the case that signaling link or signaling transfer point fails.
Signaling network functions are divided into signaling message handling & signaling network management.

Signaling Message Handling - This function ensures that signaling message originated by a particular user part at a signaling point (originating) are delivered the same user part at destination point indicated by sending user part.
Signaling Network Management - It is used to provide re-configuration of signaling network in case of failure & to control traffic in case of congestion (heavy traffic). It includes signaling traffic management, signaling link & route management.


SCCP (Signaling Connection Control Part):-
It provide additional function to MTP to cater both connection-less & connection oriented network services to transfer circuit related & non-circuit related signaling information & other type of information between switches & specialized centers in telecommunication network through SS7 network.

It is also used for transfer of circuit related & non-circuit related signaling information
of the ISDN user part with or without end to end signaling connection.

SCCP Services falls into 4 classes
1. Class 0 - Basic connectionless services.
2. Class 1 - In sequence delivery connectionless services.
3. Class 2 - Basic connection-oriented services.
4. Class 3 - Flow control connection-oriented services.

TCAP (Transaction Capabilities Application Part):-
It refers to the capabilities of providing information, request & responses. It provided functions & procedures, irrerevalent to a large variety of application between switches & databases in telecommunication network. It consists of
• TCAP - It corresponds to layer 7 of OSI model.
• ISP (Intermediate Service Part) - It corresponds to layer 4 to 6 of OSI model.

It uses SCCP supported addressing mode & is based on connection-less & connection oriented services of the SCCP.
Connection-less mode is applied in case of real-time transfer of a small amount of data.
Connection oriented mode is applied in case of non-real-time transfer of a large amount of data.


MAP (Mobile Application Part):-
This protocol defines how messages are exchanged between network entities for the purpose of realizing the MS roaming function.
The network entities involve here includes MSC, HLR, AUC (Authentication Center), MC, SCP.
In CDMA, C, D, E, T & Q interfaces all can transfer MAP messages therefore they all are referred as MAP interfaces.
MAP Functions:-
1. Location & data management - They are the basis of other services of mobile network.
It function includes
• Realizing MS automatic roaming & roaming restriction.
• Providing subscriber data for other services.
• Maintaining data consistency between HLR & VLR.
• Protecting network resources from being accessed by invalid subscribers.

2. Handoff Management - It ensures interconnection & interworking between mobile equipment of different suppliers, so that subscriber can roam freely in different MSC's. It function includes
• Basic handoff function namely forward handoff, backward handoff & handoff to third party.
• Transparent signaling transmission after handoff.
• Circuit Management.

3. Call Function - It includes origination request used to obtain calling subscriber data from the HLR or SCP.
• Location request used to obtain location information of called subscriber from HLR.
• Forwarding request to obtain forward-to request.

4. Supplementary Services - It supports various call related & non-call related supplementary services, like conference calling. The MAP can identify & support SCP oriented features operations.

5. Intelligent Network (IN) Services - It supports following IN functions
• Intelligent controls.
• SCP based forwarding services.
• Service recovery.
• In prepaid charging (PPC) service, MAP is responsible for restoring the MSCe or SCP in the case of any exception.

6. SMS Services - MAP support following functions
• MS initiating a short message.
• MS termination a short message.
• Short message broadcast.
• Short message notice.


BSAP (Base Station Application Part):-
It is the application protocol used for A interface. Interface between MSC & BSC is A interface. The A interface includes A1, A2 & A5 interface.
A1 - It transfers call control related signaling.
A2 - It transfer 64Kbps PCM voice service.
A5 - It transfer circuit-switched data service.

BSAP describes 2 types of messages -
• BS Management Application Part (BSMAP)
• Direct Transfer Application Part (DTAP)
BSMAP supports the resources management & circuit equipment management procedures between MSC & BSC.
DTAP transfers mobility management messages between BSC & MSC.
BSAP protocols defines message format & procedures to support the wireless service function between MSC & BSC. Major A interface signaling procedure includes Mobile origination, Mobile Termination, Call Clearing.


TUP (Telephone User Part):-
It defines the circuit signaling function necessary for call control, namely the content of signaling message transferred between switching offices.


ISUP (Integrated Service Digital Network User Part):-
It defines signaling messages functions & procedure required to control voice & non-voice services. It not only implements the function of TUP & Data User Part (DUP) but also realize diversified ISDN services.

The ISUP supports basic bearer services i.e. establishing, monitoring & releasing 64Kbps circuit between subscriber terminals & providing lower layer message transfer capabilities for subscriber.

It also support following supplementary services,
• Calling Line Identification Presentation & Restriction (CLIP & CLIR).
• Connected Line Identification Presentation & Restriction (COLP & CLOR).
• Call Forwarding, Call Holding, Call Waiting, User to User Signaling, Three-Way Calling, Conference Call.
ISUP also supports MultiDestination signaling point function.


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What is Signaling

In Communication system, the messages that are required to co-ordinate different entities.

Signaling messages are described in interface, protocols & specifications.

INTERFACES :- It refers to the connecting point between 2 adjacent network entities.

PROTOCOLS :- Its a set of rules to be defined for exchanging information between connecting points.

Types of Signaling :-

In telephone network, signaling is divided into 2 parts -
1. Access Signaling - Signaling between subscriber terminal (telephone) & the local exchange

2. Trunk Signaling (Inter-exchange signaling) - It is used for signaling between exchanges. Inter-exchange signaling information is usually transported on one of the time slot in PCM link, either in association with the speech channel or independently.

These are of 2 types :-

1. Channel Associated Signaling (CAS) - In CAS, speech channel (in-band) or a channel closely associated with speech channel (out-band), is used for signaling.

2. Common Channel Signaling (CCS) - In CCS, a dedicated channel completely seperate from speech channel is used for signaling. Due to high capacity, in CCS one signaling channel can serve a large no. of speech channel.

Line Signals :- They are used during the 'duration of a call' to monitor the status of the connection & traffic circuit, e.g. seizure, answers signals, etc.

Register Signals :- They are used during the setup phase of a call to transfer address & category information e.g. dailed B number, etc.

Line signals & register signals are used in CAS system.

In CCS, signaling messages (data packets) are transmitted over time slot in a PCM link reserved for the purpose of signaling. The System is designed to use a common data channel (signaling link) as the carrier of all signals required by a large no. of traffic channels.

Signaling Interfaces :-

These are A, B, C, D, zz, 39/xx, Q, T1, SIGTRAN, ISUP interface, etc.

A,B,C,D are already defined in previous Interfaces post,

ZZ:- Its between MSCe's (Mobile Switching Center Emulation). It compiles with the SIP-T (Session Initiation Protocol for Telephone). It provide the inter-office call control function for narrowband circuit-switched domain services.

39/xx:- Its between MGW (Media Gateway) & MRFP (Media Resource Function Processor). It compiles with Megaco/H.248. Its is used when MSCe controls dynamic & static resources of transmission nodes (IP/TDM) in the MGW during call processing, including terminal attributes, terminal connectivity & mobile streams.

Q:- Its between MSCe & MC (Message Center). It Compiles with MAP (Mobile Application Part) of SS7 (Signaling System 7) to support SMS (Short Message Services).

T1:- Its between MSCe & SCP (Service Control Point). It compiles with WIN of SS7 to support IN (Intelligent Network) services.

SIGTRAN:- Its between MSCe & SG (Signaling Gateway). It is used to transmit circuit switching signaling messages over IP (Internet Protocol) network.

ISUP interfaces :- The gateway office provides the interfaces between PSTN (Public Switch Telephone Network) & other mobile network devices & controls incoming & outgoing calls through ISUP or TUP of SS7.


Signaling Protocols :-
       These are

• SS7 (Including ISUP, MAP & BSAP)
• SIP (Session Initiation Protocol)
• H.248

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Erlang / Mobile Network Traffic

What is ERLANG ?
Its a dimensionless unit of traffic intensity.
One erlang is the intensity at which one traffic path i.e. one circuit would be continuously occupied.

It is equivalent of one call (including call attempts and holding time) in a specific channel for 3600 seconds in an hour. The 3600 seconds need not be, and generally are not in a contiguous block.

Example :-
Suppose 60 calls happens in one hour, each lasting 5 minutes,
Minutes of traffic in the hour = number of calls x duration = 60 x 5 = 300
Hours of traffic in the hour = 300/60 = 5
Traffic figure = 5 erlangs.

Erlang calculations are further broken down as:

Erlang B -- The most commonly used traffic model. Erlang B is used to work out, how many lines are required if the traffic figure during the busiest hour is known. This model assumes that all blocked calls are cleared immediately.

Extended Erlang B -- Similar to Erlang B, this model can be used to factor in the number of calls that are blocked and immediately tried again.

Erlang C -- This model assumes that all blocked calls are queued in the system until they can be handled. Call centers can use this calculation to determine how many call agents to staff, based on the number of calls per hour, the average duration of class and the amount of time calls are left in the queue.


Network designers use the erlang to understand traffic patterns within a voice network and use the figures to determine how many lines are required between a telephone system and a central office (PSTN exchange lines), or between multiple network locations.

Erlang is named after Danish telephone engineer A. K. Erlang.


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Intefaces among Mobile Network

What is Interface ?
            They are required to connect different nodes in the GSM network.

Types of Interfaces :-

• Air interface or Um-interface -  

 The Air Interface is the interface between the BTS (Base Transceiver Station) and the MS (Mobile Station).
The air interface is required for supporting:
- Universal use of any compatible mobile station in a GSM network.
- A maximum spectral efficiency.

• Abis-interface -

Its the interface between the BSC (Base Station Controller) and the BTS. The interface comprises traffic and control channels.
Functions implemented at the Abis-interface are:
- Voice-data traffic exchange.
- Signaling exchange between the BSC and the BTS.
- Transporting synchronization information from the BSC to the BTS.

• A-interface -

Its the interface between the BSS and the MSC. It manages the allocation of suitable radio resources to the MSs and mobility management.

• B-interface -

This interface is between between the MSC and the VLR.
VLR uses the MAP/B protocol. Most MSCs are associated with a VLR, making the B interface "internal". Whenever the MSC needs access to data regarding a MS located in its area, it interrogates the VLR using the MAP/B protocol over the B interface.

• C-interface -

It is between the HLR and a GMSC or a SMS-G.

Each call originating outside of GSM (i.e., a MS terminating call from the PSTN) has to go through a Gateway to obtain the routing information required to complete the call, and the MAP/C protocol over the C interface is used for this purpose.

• D-interface -

The D interface is between the VLR and HLR,
It uses the MAP/D protocol to exchange the data related to the location of the MS and to the management of the subscriber.

• E-interface -

It interconnects two MSCs.
The E interface exchanges data related to handover between the anchor and relay MSCs using the MAP/E protocol.

• F-interface -

It connects the MSC to the EIR (Equipment Identity Register),
It uses the MAP/F protocol to verify the status of the IMEI that the MSC has retrieved from the MS.

• G-interface -

The G interface interconnects two VLRs of different MSCs
It uses the MAP/G protocol to transfer subscriber information, e.g. during a location update procedure.

• H-interface -

The H interface is between the MSC and the SMS-G,
It uses the MAP/H protocol to support the transfer of short messages.

• I-interface -

It is the interface between the MSC and the MS.
Messages exchanged over the I interface are relayed transparently through the BSS.


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Mobile Network Elements !!!

MSC - Mobile-services Switching Center:
It’s basically an ISDN (International Subscriber Digital Number) Switch, coordinating and setting up calls to and from MSs. It performs the switching functions for all mobile stations located in the geographic area covered by its assigned BSSs.
Functions performed include interfacing with the Public Switched Telephone Network (PSTN) as well as with the other MSC’s and other system entities, such as the HLR, in the PLMN (Public Land Mobile Network).

Functions
• Call handling
• Management of required logical radio-link channel during calls
• Management of MSC-BSS signaling protocol
• Handling location registration and ensuring inter-working between Mobile Station and VLR
• Control of inter-BSS and inter-MSC handovers
• Acting as a gateway MSC to interrogate the HLR.


HLR – Home Location Register
It’s a database used to store permanent and semi-permanent subscriber data. The HLR will always know in which location area the MS is (assuming the MS is in a coverage area), and this data is used to locate an MS in the event of a MS terminating call set-up. It contains the identities of mobile subscribers called IMSIs (International Mobile Subscriber Identities), their service parameters, and their location information.

It contains,
• Identity of mobile subscriber
• ISDN directory number of mobile station
• Subscription information on teleservices and bearer services
• Service restrictions (if any)
• Supplementary services
• Location information for call routing


VLR - Visitor Location Register:
The VLR contains all the subscriber data, both permanent and temporary, which are necessary to control a MS in the MSC’s coverage area. It contains the subscriber parameters and location information for all mobile subscribers currently located in the geographical area (i.e., cells) controlled by that VLR.

It contains,
• Identity of mobile subscriber
• Any temporary mobile subscriber identity
• ISDN directory number of mobile
• A directory number to route calls to a roaming station
• Location area where the mobile station is registered
• Copy of (part of) the subscriber data from the HLR


EIR - Equipment Identity Register:
Its database contains information on the MS and its capabilities. The IMEI (International Mobile Station Equipment Identity) is used to interrogate the EIR. It is accessed during the equipment validation procedure when a mobile station accesses the system. It contains the identity of mobile station equipment which may be valid, suspect, or known to be fraudulent.

This contains:
• White or Valid list - List of valid MS equipment identities
• Grey or Monitored list - List of suspected mobiles under observation
• Black or prohibited list - List of mobiles for which service is barred


AuC - Authentication Center:
Its database contains the subscriber authentication keys and the algorithm required to calculate the authentication parameters to be transferred to the HLR.
• Contains subscriber authentication data called Authentication Keys (Ki)
• Generates security related parameters needed to authorize service using Ki
• Generates unique data pattern called a CipherKey (Kc) needed for encrypting user speech and data


OMC - Operations and Maintenance Center:
It is the centralized maintenance and diagnostic heart of the Base Station System (BSS). It allows the network provider to operate, administer, and monitor the functioning of the BSS.


BSS (Base Station Subsystem) - It’s a combination of a BSC and one or more BTSs.

Characteristics of the BSS are:
• It is responsible for communicating with mobile stations in cell areas.
• One BSC controls one or more BTSs and can perform inter-BTS and intra-BTS handovers.
• The BTS serves one or more cells in the cellular network and contains one or more TRXs (Transceivers or radio units).
• The TRX serves full duplex communications to the MS.
• In the GSM network implementation, the BSC includes the TRAU (Transcoder/Rate Adapter Unit). The TRAU adapts the transmission bit rate of the A-interface (64 kbit/s) to the Abis-interface (16 kbit/s).


MS - Mobile Station:
It represents the terminal equipment used by the wireless subscriber supported by the GSM Wireless system. The MS consists of two entities, each with its own identity:
• Subscriber Identity Module (SIM)
• Mobile equipment

The SIM may be a removable module. A subscriber with an appropriate SIM can access the system using various mobile equipments. The equipment identity is not linked to a particular subscriber. Validity checks made on the MS equipment are performed independently of the authentication checks made on the MS subscriber information.

Functions of a SIM:
• Authentication of the validity of the MS when accessing the network
• User authentication
• Storage of subscriber-related information, which can be: data fixed during administrative phase (e.g., subscriber identification), and temporary network data (e.g., cell location identity).

Functions of a Mobile Station:
• Radio transmission termination
• Radio channel management
• Speech encoding/decoding
• Radio link error protection
• Flow control of data
• Rate adaptation of user data to the radio link
• Mobility management
• Performance measurements of radio link

Types of Mobile Stations:
Mobile stations can come in different power classes, which define the maximum RF power level that the unit can transmit. For GSM-900 there is five powers classes, for GSM-1800 there are three power classes. The mobile station maximum output power is specified in GSM.

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GSM System Architecture

• Mobile-services Switching Center (MSC)·
• Home Location Register (HLR)·
• Visitor Location Register (VLR)·
• Equipment Identity Register (EIR)·
• Authentication Center (AUC)·
• Base Station System (BSS)-
• Base Transceiver Station (BTS)-
• Base Station Controller (BSC)·
• Mobile Station (MS)·
• Operation and Maintenance Center (OMC)
• Interfaces - A,B,C,D,E,F,G,Um,Abis

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