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OF TWO MONTHS INDUSTRIAL TRAINING, UNDERTAKEN AT
“REGIONAL TELECOM TRAINING CENTRE, RAJPURA”
IN “TELECOM DEPARTMENT” ON “GSM, WLL, SWITCHNING, TRANSMISSION AND NETWORKING TECHNOLOGIES” SUBMITTED IN PARTIAL FULFILLMENT OF THE DEGREE OF BACHELOR OF TECHNOLOGY IN ELECTRONICS & COMMUNICATION ENGINEERING
Submitted By: ……………………….. Name: ………………………………… College Roll No.: …………………….. University Roll No.:…………………..
Under the Guidance of: Name: M.K.BHATIA Designation: DE (ADMIN) Department: TELECOM
CHANDIGARH-PATIALA NATIONAL HIGHWAY, VILL.JHANSLA, TEHSIL, RAJPURA, DISTT. PATIALA 140401
REGIONAL TELECOM TRAINING CENTRE, RAJPURA (AN ISO 9001:2000 CERTIFIED INSTITUTE)
RTTC Rajpura was established on 01.12.75 in a rented building belonging to Kasturba Sewa Mandir Trust in Rajpura. It has been shifted at New RTTC Complex, Neelpur Village Rajpura town w.e.f. 26.7.2004. It is situated on Patiala bye pass road near Liberty Chowk. Rajpura is situated on the main line from Delhi to Amritsar at a distance of 230 Kilometers from Amritsar as well as Delhi and 30 Kilometers from Ambala. The RTTC Complex includes Academic & Administrative block, staff quarters, Inspection Quarters, Student Centre and Three Hostels. The total Trainee capacity of these Hostels is 200. The entire campus is spread over 20 acres of land. The built up area of RTTC Complex is 9700 Sq.mtrs. The campus is situated away from the town and is well suited for educational institution. The campus is extremely beautiful and the ambience rejuvenating.
SCOPE OF RTTC RAJPURA
Imparting training in the area of telecom services and information technology clauses 7.3 and Clauses 7.6 of IS/ISO 9001:2000.
RTTC Rajpura fraternity consisting of officers and staffs vows to provide quality training to all its customers and to ensure optimum utilization of its resources. For this every faculty and individual of RTTC will adhere to 9001:2000 standards and will demonstrate its compliance in all spheres of activities with a commitment to continual improvement.
To work hard to deliver all our courses as quality courses i.e. defined as having acquired 85% rating as per evaluation by the customer. To endeavor to provide excellent course content, quality presentation, handout and congenial classroom environment.
Telecommunication overview: Mobile communication, how mobile works,
technologies used in today’s communication world, how call routes.
Transmission: How signals are transmitted from one place to another in communication
process, types of media used, how they work, transmission systems.
Networking: how data is transferred from one network to another, how IP address is
configured, how a network is established, how routing of packets takes place.
Switching systems and technologies: how exchanges work, how signalling is done,
technologies and devices used in switching purposes.
Practical aspect: jointing of Optical fibre cable
A field visit to Ambala and Patiala exchanges. Joining of CAT-5 cable.
Modules covered during six weeks training program:
Topic GSM Page:5-20 MON Cell Planning
Introduction to 2-G CDMA 3-G CDMA CDMA architecture, RF planning CorDECT system and its link budget
Transmission Digital Network tech. ing Page:29-38 Page:3953 Page:5477
PCM PDH and SDH Basic CCS-7 ISO % OSI layers TCP/IP TCP/IP Routing Broadban d & CAT-5 cable joining
And Cell Structure TUE GSM architecture WED GSM architecture THU Call flow, call routing, power control FRI Call flow, call routing, power control
OFC cable and CDOT its jointing DWDM, CDOT transmission Visit to Ambala exchange OCB
Module -1 GSM TECHNOLOGY
Global System for Mobile Communication (GSM) is a set of ETSI standards specifying the infrastructure for a digital cellular service. The standard is used in approx. 85 countries in the world including such locations as Europe, Japan and Australia. GSM is worldwide standard that allows users of different operators to connect and to shares the services simultaneously. GSM has been the backbone of the phenomenal success in mobile telecommunication over the last decade. Now, at the dawn of the era of true broadband services, GSM continues to evolve to meet new demands. One of GSM's great strengths is its international roaming capability, giving consumers a seamless service in about 160 countries. This has been a vital driver in growth, with around 300 million GSM subscribers currently in Europe and Asia. In the Americas, today's 7 million subscribers are set to grow rapidly, with market potential of 500 million in population, due to the introduction of GSM 800, which allows operators using the 800 MHz band to have access to GSM technology too. The main points and strength of GSM technology is: • GSM is globally accepted standard for DIGITAL CELLULAR COMMUNICATION. • Provides recommendations and not requirements. • Defines the functions and interface requirements in detail but do not address the hardware. • It is an open interface. Why GSM? • The GSM study group aimed to provide the followings through the GSM: • Improved spectrum efficiency. • International roaming. • Low-cost mobile sets and base stations (BSs) • High-quality speech • Compatibility with Integrated Services Digital Network (ISDN) and other telephone company services. • Support for new services. • High transmission quality
GSM CELLULAR STRUCTURE
GSM uses a combination of FDMA (Frequency Division Multiple Access) and TDMA (Time division Multiple Access).The GSM system has an allocation of 50 MHz band width in 900 MHz frequency band. Now using FDMA this band is divided into 124 channels each with a carrier bandwidth of 200 KHz .Using TDMA these channels are further divided into 8 time slots. Therefore, combination of both FDMA and TDMA we can realize a maximum of 922 channels for transmit and receive .Approximately 50 KHz of RF spectrum is reserved for each subscriber and if we have a large number of subscribers(say 10000) the RF spectrum would then required will be 10000*50 KHz =5GHz. But this large Spectrum is not available for use. This limitation of RF spectrum and in order to serve a hundreds of thousands of users’ concept of frequency reuse was developed.
The frequency reuse concept lead to development of cellular technology and it was originally conceived by AT&T Bell labs in 1947.
Cell is the basic service area. The cell is the area given radio coverage by one base transceiver station. The GSM network identifies each cell via the cell global identity (CGI) number assigned to each cell. In a cellular system, the communication area of the service provider is divided into small geographical areas called cells. Each cell contains following components: • An antenna • Solar or AC power network station. • The solar or AC powered network station is called the Base Station (BS).
WHY HEXAGONAL SHAPED CELLS ARE BETTER?
Cells are drawn in hexagonal shape because the hexagonal shaped cells have no gaps or overlaps between them. It causes no interruption to the communication of a mobile subscriber moving from one cell to another. It is obvious from the figure that other shapes of the cells are leaving gaps where no coverage is provided to the mobile users. On the other hand, there is no such problem in hexagonal cells.
TYPES OF CELLS
Due to the uneven changes in the population density of different countries and regions in the world, there are different types of cells used according to the best results in uninterruptible communication. These are listed as: • Macro Cells • Micro Cells • Pico Cells • Umbrella Cells • Selective Cells
a) Macro Cells
A macro cell is a cell in a mobile phone network that provides radio coverage served by a power cellular base station (tower). Generally, macro cells provide coverage larger than micro cell such as rural areas or along highways. The antennas for macro cells are mounted on ground-based masts, rooftops and the other existing structures, at a height that provides a clear view over the surrounding buildings and terrain. Macro cell base stations have power outputs of typically tens of watts.
b) Micro Cells
A micro cell is a cell in a mobile phone network served by a low power cellular base station (tower), covering a limited area such as a mall, a hotel, or a transportation hub. A micro cell is usually larger than a Pico cell, though the distinction is not always clear. Typically the range of a micro cell is less than a mile wide. The antennas for micro cells are mounted at street level. Micro cell antennas are smaller than macro cell antennas and when mounted on existing structures c an often be disguised as building features. Micro cells provide radio coverage over distances up to, typically, between 300m and 1000m. Micro cell base stations have lower output powers than macro cells, typically a few watts.
c) Pico Cells
Pico cells are small cells whose diameter is only few dozen meters; they are used mainly in indoor applications. It can cover e.g. a floor of a building or an entire building, or for example in shopping centers or airports.  Pico cells provide more localized coverage than micro cells, inside buildings where coverage is poor or there are high numbers of users.
d) Umbrella Cells
A layer with micro cells is covered by at least one macro cell, and a micro cell can in turn cover several Pico cells, the covering cell is called an umbrella cell. If there are very small cells and a user is crossing the cells very quickly, a large number of handovers will occur among the different neighboring cells. The power level inside an umbrella cell is increased compared to the micro cells with which it is formed. This makes the mobile to stay in the same cell (umbrella cell) causing the number of handovers to be decreased as well as the work to be done by the network.
The full coverage of the cells may not be required in all sorts of applications, but cells with limited coverage are used with a particular shape. These are named selective due to the selection of their shape with respect to the coverage areas. For example, the cells used at the entrance of the tunnels are selective cells because coverage of 120 degrees is used in them.
The regular repetition of frequencies in cells results in a clustering of cells. A cluster is a group of cells. No channels are reused within a cluster. The generate in this way can consume the whole frequency band. The size of cluster is defined by k, the numbers of cells in a cluster and this also defines frequency reuse distance.
Frequency Reuse Concept
The concept of cellular systems is the use of low power transmitters in order to enable’s the efficient reuse of the frequencies. If the transmitters of high power are used, there will be interference between the users at the boundaries of the cells. However, the set of available frequencies is limited and that is why there is a need for the reuse of the frequencies.
A frequency reuse pattern is a configuration of N cells, N being the reuse factor, in which each cell uses a unique set of frequencies. When the pattern is repeated, the frequencies can be reused. There are several different patterns, but only two are shown below to clarify the idea.
A GSM system is basically designed as a combination of four major subsystems: Radio subsystem (RSS) Network subsystem (NSS) Operation and maintenance subsystem (OMS)
Radio Subsystem (RSS)
Management of radio network and is controlled by a MSC. One MSC controls many radio sub-system. The Radio Subsystem (RSS) consists of: BSC: Base station controller. BTS: Base transceiver station. Radio subsystem mainly performs following functions: • Radio path control • Synchronization • Air and A interface signaling • Connection between MS and NSS • Mobility management • Speech transcoding • Handovers
Base Station Controller (BSC)
A BSC is a network component in the PLMN that function for control of one or more BTS. It is a functional entity that handles common control functions within a BTS. BSC within a mobile network is a key component for handling and routing information. The BSC provides all the control functions and physical links between the MSC and BTS. It is a high-capacity switch that provides functions such as handover, cell configuration data, and control of radio frequency (RF) power levels in base transceiver stations. A number of BSCs are served by an MSC. The BSC is connected to the MSC on one side and to the BTS on the other. The BSC performs the Radio Resource (RR) management for the cells under its control. It assigns and releases frequencies and timeslots for all MSs in its own area. The BSC performs the inter-cell handover for MSs moving between BTS in its control. It also reallocates frequencies to the BTSs in its area to meet locally heavy demands during peak hours or on special events. The BSC controls the power transmission of both BSSs and MSs in its area. The minimum power level for a mobile unit is broadcast over the BCCH. The BSC provides the time and frequency synchronization reference signals broadcast by its BTSs. The BSC also measures the time delay of received MS signals relative to the BTS clock. If the received MS signal is not centered in its assigned timeslot at the BTS, The BSC can direct the BTS to notify the MS to advance the timing such that proper synchronization takes place. The BSC may also perform traffic concentration to reduce the number of transmission lines from the BSC to its BTSs. 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. Redundancy
often extends beyond the BSC equipment itself and is commonly used in the power supplies and in the transmission equipment providing the A-ter interface to PCU. 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.
Base Terminal Station (BTS)
The BTS handles the radio interface to the mobile station. The BTS is the radio equipment (transceivers and antennas) needed to service each cell in the network. A group of BTSs are controlled by a BSC. A BTS is a network component that serves one cell and is controlled by a BSC. BTS is typically able to handle three to five radio carries, carrying between 24 and 40 simultaneous communication. Reducing the BTS volume is important to keeping down the cost of the cell sites.
BTS with its antennae A BTS compares radio transmission and reception devices, up to and including the antennas, and also all the signal processing specific to the radio interface. A single transceiver within BTS supports eight basic radio channels of the same TDM frame.
Functions of BTS
The primary responsibility of the BTS is to transmit and receive radio signals from a mobile unit over an air interface. To perform this function completely, the signals are encoded, encrypted, multiplexed, modulated, and then fed to the antenna system at the cell site. Transcoding to bring 13-kbps speech to a standard data rate of 16 kbps and then combining four of these signals to 64 kbps is essentially a part of BTS, though; it can be done at BSC or at MSC. The voice communication can be either at a full or half rate over logical speech channel. In order to keep the mobile synchronized, BTS transmits frequency and time synchronization signals over frequency correction channel (FCCH and BCCH logical channels. The received signal from the mobile is decoded, decrypted, and equalized for channel impairments. Random access detection is made by BTS, which then sends the message to BSC. The channel subsequent assignment is made by BSC. Timing advance is determined by BTS. BTS signals the mobile for proper timing adjustment. Uplink radio channel measurement corresponding to the downlink measurements made by MS has to be made by BTS.
Network sub-system (NSS)
Performs call processing and subscriber related functions. It includes: MSC: Mobile Switching Centre HLR: Home Location Register VLR : Visitor Location Register AuC: Authentication Centre EIR: Equipment Identity Register GMSC: Gateway MSC
The network and the switching subsystem together include the main switching functions of GSM as well as the databases needed for subscriber data and mobility management (VLR). The main role of the MSC is to manage the communications between the GSM users and other telecommunication network users. The basic switching function is performed by the MSC, whose main function is to coordinate setting up calls to and from GSM users. The MSC has interface with the BSS on one side (through which MSC VLR is in contact with GSM users) and the external networks on the other (ISDN/PSTN/PSPDN). The main difference between a MSC and an exchange in a fixed network is that the MSC has to take into account the impact of the allocation of RRs and the mobile nature of the subscribers and has to perform, in addition, at least, activities required for the location registration and handover. The Network Switching Subsystem, also referred to as the GSM core network, usually refers to the circuit-switched core network, used for traditional GSM services such as voice calls, SMS, and circuit switched data calls.
There is also an overlay architecture on the GSM core network to provide packet-switched data services and is known as the GPRS core network. This allows mobile phones to have access to services such as WAP, MMS, and Internet access. All mobile phones manufactured today have both circuit and packet based services, so most operators have a GPRS network in addition to the standard GSM core network.
Mobile Switching Center (MSC)
An MSC is the point of connection to the network for mobile subscribers of a wireless telephone network. It connects to the subscribers through base stations and radio transmission equipment that control the air interface, and to the network of other MSCs and wireless infrastructure through voice trunks and SS7. An MSC includes the procedures for mobile registration and is generally co-sited with a visitor location register (VLR) that is used to temporarily store information relating to the mobile subscribers temporarily connected to that MSC. The MSC performs the telephony switching functions of the system. It controls calls to and from other telephone and data systems. It also performs such functions as toll ticketing, network interfacing, common channel signaling, and others. Other network elements of MSC
a) Billing Center
Each MSC writes call accounting records to local disk memory. Billing Center periodically polls the disk records of each MSC to collect the billing data for the PLMN.
b) Service Center
The Service Center interfaces with the MSCs to provide special services, such as the Short Message Service (SMS), to mobile subscribers in the PLMN. The Billing Center and Service Center are not a basic part of the GSM system.
Gateway MSC (G-MSC)
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.
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 GSM core 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.
Functions of HLR
The main function of the HLR is to manage the fact that SIMs and phones move around a lot. The following procedures are implemented to deal with this: Manage the mobility of subscribers by means of updating their position in administrative areas called 'location areas', which are identified with a LAC. The action of a user of moving from one LA to another is followed by the HLR with a Location area update while retrieving information from BSS as base station identity code (BSIC). Send the subscriber data to a VLR or SGSN when a subscriber first roams there. Broker between the G-MSC or SMSC and the subscriber's current VLR in order to allow incoming calls or text messages to be delivered. Remove subscriber data from the previous VLR when a subscriber has roamed away from it.
Visitor locations 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.
Data stored in VLR
• • •
IMSI (the subscriber's identity number). Authentication data. MSISDN (the subscriber's phone number). GSM services that the subscriber is allowed to access. access point (GPRS) subscribed. The HLR address of the subscriber.
Functions of VLR
• • • • • The primary functions of the VLR are: To inform the HLR that a subscriber has arrived in the particular area covered by the VLR. To track where the subscriber is within the VLR area (location area) when no call is ongoing. To allow or disallow which services the subscriber may use. To allocate roaming numbers during the processing of incoming calls. To purge the subscriber record if a subscriber becomes inactive whilst in the area of a VLR. The VLR deletes the subscriber's data after a fixed time period of inactivity and informs the HLR (e.g., when the phone has been switched off and left off or when the subscriber has moved to an area with no coverage for a long time). To delete the subscriber record when a subscriber explicitly moves to another, as instructed by the HLR.
Authentication centre (AUC)
The authentication centre (AUC) is a function to authenticate each SIM card that attempts to connect to the GSM core 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 GSM core network. If the authentication fails, then no services are possible from that particular combination of SIM card and mobile phone operator attempted. There is an additional form of identification check performed on the serial number of the mobile phone described in the EIR section below, but this is not relevant to the AUC processing. Proper implementation of security in and around the AUC is a key part of an operator's strategy to avoid SIM cloning.
Equipment Identity Register (EIR)
The EIR is a database that contains information about the identity of mobile equipment that prevents calls from stolen, unauthorized, or defective mobile stations. The AUC and EIR are implemented as stand-alone nodes or as a combined AUC/EIR node. EIR is a database that stores the IMEI numbers for all registered ME units. The IMEI uniquely identifies all registered ME. There is generally one EIR per PLMN. It interfaces to the various HLR in the PLMN. The EIR keeps track of all ME units in the PLMN. It maintains various lists of message. The database stores the ME identification and has nothing do with subscriber who is receiving or originating call. There are three classes of ME that are stored in the database, and each group has different characteristics: o White List: contains those IMEIs that are known to have been assigned to valid MS’s. This is the category of genuine equipment. o Black List: contains IMEIs of mobiles that have been reported stolen. o Gray List: contains IMEIs of mobiles that have problems (for example, faulty software, and wrong make of the equipment). This list contains all MEs with faults not important enough for barring.
OPERATION AND MAINTENANCE SUBSYSTEM (OMS)
The Operations and Maintenance Center (OMC) 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. An OMS consists of one or more Operation & Maintenance Centre (OMC) The operations and maintenance center (OMC) is connected to all equipment in the switching system and to the BSC. The implementation of OMC is
called the operation and support system (OSS). The OSS is the functional entity from which the network operator monitors and controls 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. An important function of OSS is to provide a network overview and support the maintenance activities of different operation and maintenance organizations. The OMC provides alarm-handling functions to report and log alarms generated by the other network entities. The maintenance personnel at the OMC can define that criticality of the alarm. Maintenance covers both technical and administrative actions to maintain and correct the system operation, or to restore normal operations after a breakdown, in the shortest possible time. The fault management functions of the OMC allow network devices to be manually or automatically removed from or restored to service. The status of network devices can be checked, and tests and diagnostics on various devices can be invoked. For example, diagnostics may be initiated remotely by the OMC. A mobile call trace facility can also be invoked. The performance management functions included collecting traffic statistics from the GSM network entities and archiving them in disk files or displaying them for analysis. Because a potential to collect large amounts of data exists, maintenance personal can select which of the detailed statistics to be collected based on personal interests and past experience. As a result of performance analysis, if necessary, an alarm can be set remotely. The OMC provides system change control for the software revisions and configuration data bases in the network entities or uploaded to the OMC. The OMC also keeps track of the different software versions running on different subsystem of the GSM.
MOBILE SUBSCRIBER IDENTITIES IN GSM
It would be better to discuss some of the important subscriber identities in the GSM, which make the use of this technology safer for every person whether he/she is a subscriber of GSM or not.
1) International Mobile Subscriber Identity (IMSI)
An IMSI is assigned to each authorized GSM user. It consists of a mobile country code (MCC), mobile network code (MNC) (to identify the PLMN), and a PLMN unique mobile subscriber identification number (MSIN). The IMSI is the only absolute identity that a subscriber has within the GSM system. The IMSI consists of the MCC followed by the MNC and MSIN and shall not exceed 15 digits. It is used in the case of system-internal signaling transactions in order to identify a subscriber. The first two digits of the MSIN identify the HLR where the mobile subscriber is administrated.
2) Temporary Mobile Subscriber Identity (TMSI)
A TMSI is a MSC-VLR specific alias that is designed to maintain user confidentiality. It is assigned only after successful subscriber authentication. The correlation of a TMSI to an IMSI only occurs during a mobile subscriber’s initial transaction with an MSC (for example, location updating). Under certain condition (such as traffic system disruption and malfunctioning of the system), the MSC can direct individual TMSIs to provide the MSC with their IMSI.
3) Mobile Station ISDN Number
The MS international number must be dialed after the international prefix in order to obtain a mobile subscriber in another country. The MSISDN numbers is composed of the country code
(CC) followed by the National Destination Code (NDC), Subscriber Number (SN), which shall not exceed 15 digits. Here too the first two digits of the SN identify the HLR where the mobile subscriber is administrated.
4) The Mobile Station Roaming Number (MSRN)
The MSRN is allocated on temporary basis when the MS roams into another numbering area. The MSRN number is used by the HLR for rerouting calls to the MS. It is assigned upon demand by the HLR on a per-call basis. The MSRN for PSTN/ISDN routing shall have the same structure as international ISDN numbers in the area in which the MSRN is allocated. The HLR knows in what MSC/VLR service area the subscriber is located. At the reception of the MSRN, HLR sends it to the GMSC, which can now route the call to the MSC/VLR exchange where the called subscriber is currently registered.
5) International Mobile Equipment Identity
The IMEI is the unique identity of the equipment used by a subscriber by each PLMN and is used to determine authorized (white), unauthorized (black), and malfunctioning (gray) GSM hardware. In conjunction with the IMSI, it is used to ensure that only authorized users are granted access to the system.
MS sends dialed number to BSS BSS sends dialed number to MSC 3,4 MSC checks VLR if MS is allowed the requested service. If so, MSC asks BSS to allocate resources for call. 5 MSC routes the call to GMSC 6 GMSC routes the call to local exchange of called user 7, 8, 9,10 Answer back(ring back) tone is routed from called user to MS via GMSC,MSC,BSS.
1. Calling a GSM subscribers 2. Forwarding call to GSMC 3. Signal Setup to HLR 4. 5. Request MSRN from VLR 6. Forward responsible MSC to GMSC 7. Forward Call to current MSC 8. 9. Get current status of MS 10. 11. Paging of MS 12. 13. MS answers 14. 15. Security checks 16. 17. Set up connection.
FUTURE OF GSM
o 2nd Generation GSM -9.6 Kbps (data rate) o Generation ( Future of GSM) HSCSD (High Speed Circuit Switched data) • Data rate : 76.8 Kbps (9.6 x 8 kbps) GPRS (General Packet Radio service) • Data rate: 14.4 - 115.2 Kbps EDGE (Enhanced data rate for GSM Evolution) • Data rate: 547.2 Kbps (max) 3 Generation WCDMA(Wide band CDMA) • Data rate : 0.348 – 2.0 Mbps
Module 2 WLL AND CDMA
WLL:WIRELESS LOCAL LOOP
o WLL is a system that connects subscribers to the local telephone station wirelessly. o Systems WLL is based on: o Cellular o Satellite (specific and adjunct) o Microcellular o Other names o Radio In The Loop (RITL) o Fixed-Radio Access (FRA).
A GENERAL WLL SETUP
HISTORY OF WLL
Wireless access first started to become a possibility in the 1950s and 1960s as simple radio technology reduced in price. For some remote communities in isolated parts of the country, the most effective manner of providing communication was to provide a radio, kept in a central part of the community. By the end of the 1970s, communities linked by radio often had dedicated radio links to each house, the links connected into the switch such that they were used in the same manner as normal twisted-pair links. The widespread deployment of the cellular base station into switching sites helped with cost reduction. Similar access using pointto-point microwave links still continues to be widely used today. During the reunification of West and East Germany, much funding was put into increasing the teledensity in East Germany. The installation of twisted-pair access throughout would have been a slow process. In the interim, cellular radio was seen to offer a stop-gap measure to provide rapid telecommunications capability. So in East Germany a number of cellular networks, based upon the analog Nordic Mobile Telephone (NMT) standard, were deployed in the 800 MHz frequency range. The key difference was that subscribers had fixed unit mounted to the sides of their houses to increase the signal strength and hence allow the networks to be constructed with larger cells for lower costs. Thus, we see the first WLL network was born. Early 1950s. Single-channel VHF subscriber equipment was purchased from Motorola, but the maintenance costs were too high as a result of the valve technology used and the power consumption too high. The trial was discontinued and the subscribers were connected by wire Mid-1950s. Raytheon was given seed funds to develop 6 GHz band equipment, which would have a better reliability and a lower power consumption. The designers failed to achieve those goals and the system still proved too expensive Late1950s. Some equipment capable of providing mobile service to rural communities was put on trial. Users were prepared to pay a premium for mobile use, but the system still proved to be too expensive in a fixed application for which users were not prepared to pay a premium. EarlY1960s. Systems able to operate on a number of radio channels were developed, eliminating the need for each user to share a specific channel and thus increase capacity. The general lack of channels and high cost, however, made these systems unattractive. Early 1970s. A Canadian manufacturer developed equipment operating at 150 MHz that proved successful in serving fixed subscribers on the island of Lake Superior. The lack of frequencies in the band, however, precluded its widespread use. Late 1970s. The radio equipment from several US manufacturers was linked to provide service to isolated Puerto Rican villages. The service was possible only because the geographical location allowed the use of additional channels, providing greater capacity than would have been possible elsewhere. Early 1980s. Communication satellites were examined for rural applications but were rejected as being too expensive.
1985. Trials of a point-to-multipoint radio system using digital modulation promised sufficient capacity and reliability to make WLL look promising
WLL VERSUS WIRELINE
The cost of installing or maintain wireline systems broadly depends on the cost of labour
whereas cost of wireless depends on cost of subscriber unit, which tends to fall over time, with increasing economics of scale. Cost of wireline is more due to use of copper a costly metal U/G cables whereas WLL is independent of this factor. Cost of wireline critically depends upon distances between houses and penetration levels achieved. These factors are not there in the WLL. If a subscriber moves to a different operator in case of wireline system, investment is lost whereas in case of WLL the subscriber unit in such a case is simply removed and installed elsewhere. The cost of wireline system is incurred, even prior to marketing to the users whereas much of cost of WLL is not incurred until the users subscribe the Network.
• • •
The widely used WLL systems are: 1. CorDECT WLL 2. CDMA WLL
MOBILE CELLULAR SYSTEM
ANALOG CELLULAR RADIO TECNOLOGIES
There is significant momentum to use analog cellular for WLL because of its wide availability resulting from serving high-mobility markets. There are three main analog cellular system types operating in the world: advanced mobile phone system (AMPS), Nordic mobile telephone (NMT), and total access communications systems (TACS). As a WLL platform, analog cellular has some limitations in regards to capacity and functionality. In the late 1990's Analog cellular systems were expected to be the major wireless platform for WLL.
DIGITAL CELLULAR TECNOLOGIES
Digital cellular has seen rapid growth and has outpaced analog cellular over the last few years. Major worldwide digital cellular standards include global system for mobile communications (GSM), time-division multiple access (TDMA), Hughes enhanced TDMA (E–TDMA), and code-division multiple access (CDMA). Although GSM is a dominant mobile digital cellular standard, there has been little activity in using GSM as a WLL platform. It offers higher capacity than the other digital standards (more than 10 to 15 times greater than analog cellular), relatively high-quality voice, and a high level of privacy. Digital cellular is expected to play an important role in providing WLL because it, like analog cellular, has the benefit of wide availability. Digital cellular can support higher capacity subscribers than analog cellular, and it offers functionality that is better suited to emulate capabilities of advanced wireline networks. Approximately one-third of the installed WLLs were using digital cellular technology by year 2000.
DRAWBACKS OF MOBILE TELEPHONY:
• • • • Lower voice quality. Higher call blocking rate. Limited subs density. Expensive.
CORDLESS TECNOLOGIES: CT-2(cordless telephony 2nd generation) • DECT(digitally enhanced cordless telephony) • PHS(phony handy system)
corDECT Wireless in Local Loop System is based on Digital Enhanced Cordless Telecommunications (DECT) standard of European Telecommunications standards Institute (ETSI).
The CorDECT Wireless in Local Loop has been designed to be a modular system. The basic unit provides service to 10o00 subscriber. Multiple CorDECT systems can be connected together using a transit switch. The system has been designed in such a way that the initial investment for fixed part is very low and most of the cost is incurred on the Subscriber Unit, which needs to be obtained only when the operator signs up a subscriber. Further since CorDECT Wireless in Local Loop does not require frequency planning the installation need not be coordinated. Thus the low cost marks the system one of the most versatile in Local Loop System available today.
The CorDECT system is designed to provide a cost effective wireless high quality voice and DATA connection in dense Urban as well as sparse rural areas. The system enables wireless subscriber to be connected to the PSTN in a cost effective manner.
DECT INTERFACE UNIT (DIU)
The DIU is a dect exchanges for Wireless subscriber and provides an interface to a public Switched Telephone Network (PSTN). Functions such as call processing, CBS powering and PCM/ADPCM transcoding. DECT Network Layer and link Layer functioned are handled by the DIU. System operation and Maintenance (O&M) and remote fault monitoring can be performed from the DIU or alternatively from a remote location using the Network Management System. For 1000 wireless subscriber the DIU can be configured as – a) b) c) An exchange with R2-MF signaling on EI lines or A Remote Switching Unit (RSU) to an exchange using V5.2 protocols on EI lines or, An in dialing PBX connected to an exchange using two-wire junction lines or connected to PSTN using R2-MF signaling on EI lines. An optional subscriber MUX (SMUC) unit in the DIU converts the EI interface to 30 junctions lines, which can be connected to two-wire subscriber lines of an exchange. The SMUX carries out polarity reversal detection and 16 KHz metering pulse detection. It allows pulse dialing and DTMF dialing. The two-wire state is coded a transmitted on slot 16 of the EI line.
The DIU consists of between three to six standard 19” sub-racks in one or two cabinets depending on the CorDECT system configuration. All critical cards have a hot standby so that system availability is ensured in case of failure. The system is powered by 48V power supply.
COMPACT BASE STATION (CBS)
It provides wireless access in an area and supports twelve simultaneous full duplex channels. The CBS is a small unobtrusive pole mounted or wall mounted unit. Each CBS serves one cell providing upto 12 simultaneous speech channel gain of the handset/ wall set. Typically it ranges from 150m-5kms. (10kms in rural areas). The CBS has two antenna for diversity. A direction antenna with significant gain can be used when coverage required is either confined to certain directions, or the coverage area is divided into sector covered by different CBSs. Otherwise an omni-directional antenna could be used. Such omni-directional antenna with 2 db, 4db, and 6db gain is available. The CBS interfaced to the DIU using 3 standard subscriber pair from the existing loop plant. Typically this would be from the reliable buried portion of the loop plant terminating at the distribution points. The three pairs carry four ADPCM speech channel each in addition to signaling data 2 B+D format of N-ISDN communications. The pairs also supply power to the CBS from the DIU. The maximum distance between CBS and DIU is 4kms. With 0.4 mm dia copper twisted pairs. Alternatively the CBSs are interfaced the DIU through the base station distributor (BSD) unit as shown in fug… In the case of BSD is connection to a DIU with an EI link using radio or fibre and CBS are connected to the BSD using three pairs of twisted pair copper able each of which carries both the power as well as signal to the CBS. The maximum distance between CBSW and BSD is 1km when 0.6mm twisted pair copper cable is used.
BASE STATION DISTRIBUTOR (BSD)
The base station distributor is an optional unit used when a cluster of the CBSS are to be located some distance away from the DIU. The BSD is connected to the DIU on EI lines & each EI line carrier signals for four CBSs. The BSD demultiplexes the signal on the EI line and fides it to the four CBS. The four CBS are connected to the BSD each using 3 pairs of 0.6mm twisted pair copper wires. The maximum distance supported is 1km. The copper wires carry both power and signals from BSD to CBS The health of the BSD as w4ellas the CBSs can be upgraded from the DIU.
The wall set is a small wall-mounted unit with an external antenna and powered from A/C Mains. An internal battery provides backup in case of power failure. The external antenna provides gain and extended the range of a CBS in areas where CBS densidity is low the wall set provides a standard RJ-11 telephone socket so that any telephone FAX Machine modem or even a payphone can be connected to it. The data rate supported on modem is typically 9600 kbph as the voice is code (32 kbps ADPCM)_ before transmission on air.
The wall set software includes modem software DECT MAC layer, link layer Network layer and IWU layer software. ADPCM transcoding and DTMF tone detection are also implemented
in software, in future a V.35 /RS232 interface will be provided at the wall set so that a PC can be connected to it without a modem.
Switching in telecommunication is defined as the transfer of call from one user to another.
Circuit switching is the transmission technology that has been used since the first communication networks in the nineteenth century. In circuit switching, a caller must first establish a connection to a called person before any communication is possible. During the connection establishment, resources are allocated between the caller and the called. Generally, resources are frequency intervals in a Frequency Division Multiplexing (FDM) scheme or more recently time slots in a Time Division Multiplexing (TDM) scheme. The set of resources allocated for a connection is called a circuit, as depicted. A path is a sequence of links located between nodes called switches. The path taken by data between its source and destination is determined by the circuit on which it is flowing, and does not change during the lifetime of the connection. The circuit is terminated when the connection is closed.
Basic principle of electronic exchange
The basic purpose of an exchange is to provide temporary path for simultaneous ,bi-directional transmission of speech between. (i) Subscriber lines connected to same exchange (local exchange) (ii) Subscriber lines and trunks to other exchange(outgoing trunk call) (iii) Subscriber lines and trunks from other exchange(incoming trunk call)
(iv)Pairs of trunk towards different exchanges (transit switching)
These are also called switching functions of exchange and are implemented through equipment called switching functions. These are also called the switching functions of an exchange and are implemented through the equipment called the switching network. An exchange, which can setup just the first three types of connections is called a Subscriber or Local Exchange. If an exchange can setup only the fourth type of connections, it is called a Transit or Tandem Exchange. The other distinguished functions of an exchange are:
Storage program control exchange (SPC)
In a SPC exchanges a processor similar to general purpose computer, is used to control functions of exchange. All control functions are represented by series of various instructions is
stored in memory. Therefore processor memory holds all exchanges dependent data such as subscriber date, translation tables, routing and charging information and call records. In an SPC exchange, all control equipment can be replaced by a single processor. The processor must, therefore, be quite powerful; typically, it must process hundreds of calls per second, in addition to performing other administrative and maintenance tasks. However,
totally centralized control has drawbacks. The software for such a central processor will be voluminous, complex, and difficult to develop reliably. Moreover, it is not a good arrangement from the point of view of system security, as the entire system will collapse with the failure of the processor. These difficulties can be overcome by decentralizing the control. Some routine functions, such as scanning, signal distributing, marking, which are independent of call processing, can be delegated to auxiliary or peripheral processors. These peripheral units, each with specialized function, are often themselves controlled by small stored programs processors, thus reducing the size and complexity at central control level. Since, they have to handle only one function, their programs are less voluminous and far less subjected to change than those at central. Therefore, the associated program memory need not be modifiable (generally, semiconductors ROM's are used).
Basic schematic of SPC exchange
SPC exchanges consist of six main subsystems 1. 2. 3. 4. 5. 6. Terminating equipment Switching network Switching processor Switching peripherals Signaling interfaces Data processing peripherals
In this equipment, line, trunk, and service circuits are terminated for detection, signaling, speech transmission, and supervision of calls. In contrast to electromechanical circuits, the Trunk and Service circuits in SPC exchanges are considerably simpler because functions, like counting, pulsing, timing charging, etc. are delegated to stored program.
In an electronic exchange, the switching network is one of the largest sub-system in terms of size of the equipment. Its main functions are It mainly performs two main functions:Switching i.e. setting up temporary connection between two or more exchange terminations for transmission of speech and signal between these terminations.
TYPES OF SWITCHING NETWORK
There are two types of electronic switching systems Viz. Space Division and Time Division
Space division switching system
In it a continuous physical path is set up between i/p and o/p terminations and this path is separate for each connection and is held for entire duration of call .path for different connections is independent of each other. Once a continuous is established, signals are interchanged between two terminations. They have advantage of compatibility with existing line and trunk signaling conditions network.
Time division switching system
In it a number of connections share the same path on time division sharing basis. Path is not separate for each call but it is shared sequentially for a fraction of time by different calls. This process is repeated periodically at suitable high rate. The repetition rate is 8 KHz i.e. once every 125 microseconds for transmitting speech on network by many calls. The Time Division Switching was initially accomplished by Pulse Amplitude Modulation (PAM). With the advent of Pulse Code Modulation (PCM), the PAM signals were converted into a digital format overcoming the limitations of analog and PAM signals. PCM signals are suitable for both transmission and switching. The PCM switching is popularly called Digital Switching.
iii) Switching processor
The Switching processor is a special purpose real time computer, designed and optimized for dedicated applications of processing telephone calls. It has to perform some functions like reception of dialed digits, sending of digits in case of transit exchange.
Central control processor: - It is a high speed data processing unit, which controls the
operation of switching network. It mainly controls the three sections as shown.
Program store: - In it set of instructions called programs are stored. These programs are interpreted and executed by the central control. Translation store: - It contains information regarding lines. E.g. category of calling and called line, routing code, charging information, etc.
Data store: - It provides space for temporary storage of transient data, required in
processing telephone calls, such as digits dialed by subscriber, busy/idle state of lines and trunks, etc.
Switching peripheral equipment
The time interval in which switching processor operates is in order of microsecond where as components in telephone operates in milliseconds. The interface used to connect them is known as switching peripheral equipment The various equipments used are:-
Its purpose is to detect and inform CC of all significant events / signals on subscriber lines and trunks connected to the exchange. These signals may either be continuous or discrete. The equipments at which the events / signals must be detected are equally diverse. i. Terminal equipment for subscriber lines and inter-exchange trunks and.
Common equipment such as DTMF (Dual - Tone Multi Frequency) or MFC digit receivers and inter-exchange signaling senders / receivers connected to the lines and trunks.
To detect new calls, while complying with the dial tone connection specifications, each line must be scanned about every 300 milliseconds. It means that in a 40,000 lines exchange (normal size electronic exchange) 5000 orders are to be issued every 300 milliseconds, assuming that eight lines are scanned simultaneously.
Marker performs physical setup and release of paths through the switching network, under the control of CC. A path is physically operated only when it has been reserved in the central control memory. Similarly, paths are physically released before being cleared in memory, to keep the memory information updated vis-à-vis switching network. Depending upon whether switching is Time division or Space division, marker either writes information in the control memory (Time Division Switching), or physical operates the cross - points (Space Division Switching).
It is a buffer between high - speed - low - power CC and relatively slow-speed-high-power signaling terminal circuits. A signal distributor operates or releases electrically latching relays in trunks and service circuits, under the direction of central control.
Various switching peripherals are connected to the central processor by means of a common system. A bus is a group of wires on which data and commands pulses are transmitted between the various sub- units of a switching processor or between switching processor and switching peripherals. The device to be activated is addressed by sending its address on the address bus. The common bus system avoids the costly mesh type of interconnection among various devices.
Line Interface Circuits
To enable an electronic exchange to function with the existing outdoor telephone network, certain interfaces are required between the network and the electronic exchange.
Analogue Subscriber Line Interface
The functions of a Subscriber Line Interface, for each two-wire line, are often known by the acronym: BORSCHT B: O: R: Battery feed Overload protection Ringing
S: C: H: T:
Supervision of loop status Codec Hybrid Connection to test equipment
All these functions cannot be performed directly by the electronic circuits and, therefore, suitable interfaces are required. v) Transmission Interface Transmission interface between analogue trunks and digital trunks (individual or multiplexed) such as, A/D and D/A converters, are known as CODEC, These may be provided on a per-line and per-trunk basis or on the basis of one per 30 speech channels.
A typical telephone network may have various exchange systems (Manual, Stronger, Cross bar, Electronic) each having different signaling schemes. In such an environment, an exchange must accommodate several different signaling codes.
In common channel signaling technique, all the signaling information for a number of calls is sent over a signaling link independent of the inter-exchange speech circuits. Higher transmission rate can be utilized to enable exchange of much larger amount of information. This results in faster call setup, introduction of new services, e.g.., abbreviated dialing, and more retrials ultimately accomplishing higher call completion rate, Moreover, it can provide an efficient means of collecting information and transmitting orders for network management and traffic engineering.
Data Processing Peripherals.
Following basic categories of Data Processing Peripherals are used in operation and maintenance of exchange.
Man - machine dialogue terminals, like Tele-typewriter (TTY) and Visual Display Units (VDU), are used to enter operator commands and to give out lowvolume data concerning the operation of the switching system. These terminals may be local i.e. within a few tense of meters of the exchange, or remotely located. These peripherals have been adopted in the switching Systems for their ease and flexibility of operation.
Special purpose peripheral equipment is, sometimes employed for carrying out
repeated functions, such as, subscriber line testing, where speed is more important than flexibility.
High speed large capacity data storage peripherals (Magnetic Tape Drives,
magnetic Disc Unit) are used for loading software in the processor memory.
Maintenance peripherals, such as, Alarm Annunciations and Special Consoles, are
used primarily to indicate that automatic maintenance procedure have failed and manual attention is necessary
C- DOT (CENTER OF DEVELOPMENT OF TELEMATICS)
C-DOT DSS MAX is a universal digital switch which can be configured for different applications as local, transit, or integrated local and transit switch. High traffic/load handling capacity upto 8,00,000 BHCA with termination capacity of 40,000 Lines as Local Exchange or 15,000 trunks as Trunk Automatic Exchange. C.DOT:- Center of development of telematics. DSS :Digital switching system. MAX :- Main automatic exchange
OBJECTIVES OF C.DOT
1. Work on telecom technology products and services. 2. Provide solutions for current and future requirements of telecommunication and converged networks including those required for rural application. 3. Provide market orientation to R & D activities and sustain C-DOT as center of excellence. 4. Build partnerships and joint alliances with industry , solution provides, telcoms and other development organizations to offer cost effective solution .
ARCHITECTURE OF C-DOT DSS MAX
C-DOT DSS MAX exchanges can be configured using four basic modules. 1. Base Module 2. Central Module 3. Administrative Module 4. Input Output Module
1 Base module:
The Base Module is the basic growth unit of the system . It interfaces the external world to the switch. The interfaces may be subscriber lines, Along and digital trunks. Each Base Module can interface up to 2024 terminations. The number of Base Modules directly corresponds to the exchange size. It carries out majority of call processing function and in a small exchange application, it also carries out operation and maintenance function with the help of Input-Output Module.
The basic functions of base module are as follow :• Analog to digital and digital to analog conversion. • Interface to digital trunks and digital subscribers. • Communication to AM module via Central module. • Provision of special circuits like digital tones, announcements, etc. • Switching the call between terminals connected to same base module.
Hardware architecture of base module
Analog Terminal Unit Analog Terminal Unit consists of terminal cards, which may be a combination of Analog Subscriber Line Cards, Analog Trunk card & some Special Service Cards. i) Analog Subscriber Ling Cards. ii) Analog Trunks Cards . iii) Signaling Processor Cards . iv) Terminal interface controller (TIC) Cards . v) Special Service Cards .
Digital Terminal Unit –
Digital Terminal Unit (DTU) is used exclusively to interface digital trunks. One set of Digital Trunk Synchronization (DTS) card alongwith the Digital Trunk Controller (DTC) card is used to provide one E-1 interface. Each interface occupies one TG of 32 channels and four such interfaces share 4 TGs in a Digital Terminal Unit.
Signalling Unit Module (SUM)
• • Any one of the ATU or DTU in a BM can be replaced by SUM frame to support CCS7 signalling. Only one such unit is equipped in the exchange irrespective of its configuration or capacity.
ISDN Terminal Unit
• • • One of the four ATUs/DTUs in a BM can be replaced by ISTU to provide BRI/PRI interfaces in C-DOT DSS. The only constraint is that ISTU has to be principal TU i.e. directly connected to TSU on 8 Mbps PCM link. By equipping one ISTU in the exchange, a max. of 256 B channels are available to the administrator which can be configured as BRI, PRI or any mix as per site requirement.
Time Switch Unit –
Time Switch Unit (TSU) implements three basic functions As time switching within the Base Module, routing of control-messages within the Base Module and across Base Modules and support services like MF/DTMF circuits, answering circuits, tones, etc. These functions are performed by three different functional units, integrated as time switch unit in a single frame.
Service Unit (SU)
Service Unit is integrated around three different cards as Tone Generator with Answering Circuit (TGA), Service Circuit Interface Controller (SCIC) and MF/DTMF Controller (MFC) Card. Two MFC cards are grouped to form a terminal group. Upto four MFC Cards can be equipped
Base Message Switch (BMS)
• • • Base Message Switch (BMS) routes the control messages within the Base Module, across different Base Modules, and also Administrative Module via the Central Module.
It is implemented around two different cards as Message Switch Controller (MSC) with six direct HDLC-links and the Message Switch Device (MSD) Card implementing 16 switched HDLC links. As a unit, total 22 HDLC channels are implemented for communication with the Base Processor To support 8,00,000 BHCA, MSC and MSD cards are replaced by a High performance Message Switch (HMS) with high speed, 32 bit microprocessor (MC 68040). It implements 38 HDLC links
Time Switch (TS)
The Time Switch complex is implemented using three different functional cards as multiplexer/demultiplexer (TSM), time switch (TSS) and time switch controller (TSC). The Time Switch complex receives the following PCM links and performs time- switching on them for switching within the Base Module: (1) Four 128-channel multiplexed links from four different Terminal Units which may be any combination of ATU, DTU, #7SU and ISTU. (2) One 128-channel multiplexed BUS from the Service Circuits Interface Controller (SCIC) in the Time Switch Unit.
(3)Three 128-channel links to support onboard three party conference circuits (3 x 128).
Base Processor Unit –
• Base Processor Unit (BPU) is the master controller in the Base Module. • It is implemented as a duplicated controller with memory units. • These duplicated sub-units are realized in the form of the following cards : • Base Processor Controller (BPC) Card • Base Memory Extender (BME) Card
MODULE 4 TRANSMISSION
An optical fiber (or fiber) is a glass or plastic fiber that carries light along its length. Fiber optics is the overlap of applied science and engineering concerned with the design and application of optical fibers. Optical fibers are widely used in fiber-optic communications, which permits transmission over longer distances and at higher bandwidths (data rates) than other forms of communications. Fibers are used instead of metal wires because signals travel along them with less loss, and they are also immune to electromagnetic interference. Fibers are also used for illumination, and are wrapped in bundles so they can be used to carry images, thus allowing viewing in tight spaces. Specially designed fibers are used for a variety of other applications, including sensors and fiber lasers. Light is kept in the core of the optical fiber by total internal reflection. This causes the fiber to act as a waveguide. Fibers which support many propagation paths or transverse modes are called multi-mode fibers (MMF), while those which can only support a single mode are called single-mode fibers (SMF). Multi-mode fibers generally have a larger core diameter, and are used for short-distance communication links and for applications where high power must be transmitted. Single-mode fibers are used for most communication links longer than 550 meters (1,800 ft). Joining lengths of optical fiber is more complex than joining electrical wire or cable. The ends of the fibers must be carefully cleaved, and then spliced together either mechanically or by fusing them together with an electric arc. Special connectors are used to make removable connections.
HISTORY OF OPTICAL FIBRE:
OPTICAL FIBRE BY DANIEL COLLADON
OPTICAL FIBRE COMMUNICATION
With the development of extremely low-loss optical fibers during the 1970s, optical fiber communication became a very important form of telecommunication almost instantaneously. For fibers to become useful as light waveguides (or light guides) for communications applications, transparency and control of signal distortion had to be improved dramatically and a method had to be found to connect separate lengths of fiber together. Optical fiber can be used as a medium for telecommunication and networking because it is flexible and can be bundled as cables. It is especially advantageous for long-distance communications, because light propagates through the fiber with little attenuation compared to electrical cables. This allows long distances to be spanned with few repeaters. Additionally, the per-channel light signals propagating in the fiber can be modulated at rates as high as 111 gigabits per second, although 10 or 40 Gb/s is typical in deployed systems. Each fiber can carry many independent channels, each using a different wavelength of light (wavelength-division multiplexing (WDM)) The transparency objective was achieved by making glass rods almost entirely of silica. These rods could be pulled into fibers at temperatures approaching 3600°F (2000°C). Reducing distortion over long distances required modification of the method of guidance employed in early fibers. These early fibers (called step-index fibers) consisted of two coaxial cylinders (called core and cladding) which were made of two slightly different glasses so that the core glass had a slightly higher index of refraction than the cladding glass. By reducing the core size and the index difference in a step-index fiber, it is possible to reach a point at which only axial propagation is possible. In this condition, only one mode of propagation exists. These single-mode fibers can transmit in excess of 1011 pulses per second over distances of several hundred miles.
The problem of joining fibers together was solved in two ways. For permanent connections, fibers can be spliced together by carefully aligning the individual fibers and then employing or fusing them together. For temporary connections, or for applications in which it is not desirable to make splices, fiber connectors have been developed. Over short distances, such as networking within a building, fiber saves space in cable ducts because a single fiber can carry much more data than a single electrical cable. Fiber is also immune to electrical interference; there is no cross-talk between signals in different cables and no pickup of environmental noise. Non-armored fiber cables do not conduct electricity, which makes fiber a good solution for protecting communications equipment located in high voltage environments such as power generation facilities, or metal communication structures prone to lightning strikes. They can also be used in environments where explosive fumes are present, without danger of ignition. Wiretapping is more difficult compared to electrical connections, and there are concentric dual core fibers that are said to be tap-proof.
Although fibers can be made out of transparent plastic, glass, or a combination of the two, the fibers used in long-distance telecommunications applications are always glass, because of the lower optical attenuation. Both multi-mode and single-mode fibers are used in communications, with multi-mode fiber used mostly for short distances, up to 550 m (600 yards), and single-mode fiber used for longer distance links. Because of the tighter tolerances required to couple light into and between single-mode fibers (core diameter about 10
micrometers), single-mode transmitters, receivers, amplifiers and other components are generally more expensive than multi-mode components.
In principle, any light source could be used as an optical transmitter. In modern optical communication systems, however, only lasers and light-emitting diodes are generally considered for use. The simplest device is the light-emitting diode which emits in all directions from a fluorescent area located in the diode junction. Since optical communication systems usually require well-collimated beams of light, light-emitting diodes are relatively inefficient. On the other hand, they are less expensive than lasers and, at least until recently, have exhibited longer lifetimes. Another device, the semiconductor laser, provides comparatively well-collimated light. In this device, two ends of the junction plane are furnished with partially reflecting mirror surfaces which form an optical resonator. As a result of cavity resonances, the light emitted through the partially reflecting mirrors is well collimated within a narrow solid angle, and a large fraction of it can be captured and transmitted by an optical fiber.Both light-emitting diodes and laser diodes can be modulated by varying the forward diode current.
Semiconductor photodiodes are used for the receivers in virtually all optical communication systems. There are two basic types of photodiodes in use. The most simple comprises a reversebiased junction in which the received light creates electron-hole pairs. These carriers are swept out by the electric field and induce a photocurrent in the external circuit. The minimum amount of light needed for correct reconstruction of the received signal is limited by noise superimposed on the signal by the following circuits.
Avalanche photodiodes provide some increase in the level of the received signal before it reaches the external circuits. They achieve greater sensitivity by multiplying the photo generated carriers in the diode junction. This is done by creating an internal electric field sufficiently strong to cause avalanche multiplication of the free carriers.
ADVANTAGES OF OPTICAL FIBRE COMMUNICATION
o o o o o o o o o o o o o o o o High information carrying capacity. Resources plentiful. Less attenuation. Greater safety. Immunity to radio frequency interference. Immunity to electromagnetic interference. No cross talk is there. Small size and light weight. Less temperature sensitive. Low cost More reliable. Signal security. Small size and weight. Ruggedness and flexibility. Enormous bandwidth. Low transmission loss.
CLASSIFICATION OF OPTICAL FIBRE:
Fibers are classified according to the number of modes that they can propagate. Single mode fibers can propagate only the fundamental mode. Multimode fibers can propagate hundreds of modes. However, the classification of an optical fiber depends on more than the number of modes that a fiber can propagate. An optical fiber's refractive index profile and core size further distinguish single mode and multimode fibers. The refractive index profile describes the value of refractive index as a function of radial distance at any fiber diameter. Fiber refractive index profiles classify single mode and multimode fibers as follows: • Multimode step-index fibers • Multimode graded-index fibers • Single mode step-index fibers • Single mode graded-index fibers
PROPAGATION OFLIGHT THROUGH A MULTIMODE FIBRE
STEP-INDEX MULTIMODE FIBRE
In a step-index multi-mode fiber, rays of light are guided along the fiber core by total internal reflection. Rays that meet the core-cladding boundary at a high angle (measured relative to a line normal to the boundary), greater than the critical angle for this boundary, are completely reflected. The critical angle (minimum angle for total internal reflection) is determined by the difference in index of refraction between the core and cladding materials. Rays that meet the boundary at a low angle are refracted from the core into the cladding, and do not convey light and hence information along the fiber. The critical angle determines the acceptance angle of the fiber, often reported as a numerical aperture. A high numerical aperture allows light to propagate down the fiber in rays both close to the axis and at various angles, allowing efficient coupling of light into the fiber. However, this high numerical aperture increases the amount of dispersion as rays at different angles have different path lengths and therefore take different times to traverse the fiber. A low numerical aperture may therefore be desirable.
GRADED INDEX MULTIMODE FIBRES
In graded-index fiber, the index of refraction in the core decreases continuously between the axis and the cladding. This causes light rays to bend smoothly as they approach the cladding, rather than reflecting abruptly from the core-cladding boundary. The resulting curved paths reduce multi-path dispersion because high angle rays pass more through the lower-index periphery of the core, rather than the high-index center. The index profile is chosen to minimize the difference in axial propagation speeds of the various rays in the fiber. This ideal
index profile is very close to a parabolic relationship between the index and the distance from the axis.
THE STRUCTURE OF A SINGLE MODE FIBER
Fiber with a core diameter less than about ten times the wavelength of the propagating light cannot be modeled using geometric optics. Instead, it must be analyzed as an electromagnetic structure, by solution of Maxwell's equations as reduced to the electromagnetic wave equation. The electromagnetic analysis may also be required to understand behaviors such as speckle that occur when coherent light propagates in multi-mode fiber. As an optical waveguide, the fiber supports one or more confined transverse modes by which light can propagate along the fiber.
SINGLE MODE STEP INDEX FIBER
Fiber supporting only one mode is called single-mode or mono-mode fiber. The most common type of single-mode fiber has a core diameter of 8–10 micrometers and is designed for use in the near infrared. The mode structure depends on the wavelength of the light used, so that this fiber actually supports a small number of additional modes at visible wavelengths. has a narrow core (eight microns or less), and the index of refraction between the core and the cladding changes less than it does for multimode fibers. Light thus travels parallel to the axis, creating little pulse dispersion. Telephone and cable television networks install millions of kilometers of this fiber every year.
OPTICAL FIBRE CABLES
DESIGN OF OPTICAL FIBRE COMMUNICATION
In practical fibers, the cladding is usually coated with a tough resin buffer layer, which may be further surrounded by a jacket layer, usually plastic. These layers add strength to the fiber but do not contribute to its optical wave guide properties. Rigid fiber assemblies sometimes put
light-absorbing ("dark") glass between the fibers, to prevent light that leaks out of one fiber from entering another. This reduces cross-talk between the fibers, or reduces flare in fiber bundle imaging applications. For indoor applications, the jacketed fiber is generally enclosed, with a bundle of flexible fibrous polymer strength members like Aramid (e.g. Twaron or Kevlar), in a lightweight plastic cover to form a simple cable. Each end of the cable may be terminated with a specialized optical fiber connector to allow it to be easily connected and disconnected from transmitting and receiving equipment.
For use in more strenuous environments, a much more robust cable construction is required. In loose-tube construction the fiber is laid helically into semi-rigid tubes, allowing the cable to stretch without stretching the fiber itself. This protects the fiber from tension during laying and due to temperature changes. Loose-tube fiber may be "dry block" or gel-filled. Dry block offers less protection to the fibers than gel-filled, but costs considerably less. Instead of a loose
tube, the fiber may be embedded in a heavy polymer jacket, commonly called "tight buffer" construction. Tight buffer cables are offered for a variety of applications, but the two most common are "Breakout" and "Distribution". Breakout cables normally contain a rip cord, two non-conductive dielectric strengthening members (normally a glass rod epoxy), an aramid yarn, and 3 mm buffer tubing with an additional layer of Kevlar surrounding each fiber. Distribution cables have an overall Kevlar wrapping, a ripcord, and a 900 micrometer buffer coating surrounding each fiber. These fiber units are commonly bundled with additional steel strength members, again with a helical twist to allow for stretching. A critical concern in cabling is to protect the fiber from contamination by water, because its component hydrogen (hydronium) and hydroxyl ions can diffuse into the fiber, reducing the fiber's strength and increasing the optical attenuation. Water is kept out of the cable by use of solid barriers such as copper tubes, water-repellant jelly, or more recently water absorbing powder, surrounding the fiber. Finally, the cable may be armored to protect it from environmental hazards, such as construction work or gnawing animals. Undersea cables are more heavily armored in their near-shore portions to protect them from boat anchors, fishing gear, and even sharks, which may be attracted to the electrical power signals that are carried to power amplifiers or repeaters in the cable. Modern fiber cables can contain up to a thousand fibers in a single cable, so the performance of optical networks easily accommodates even today's demands for bandwidth on a point-topoint basis. However, unused point-to-point potential bandwidth does not translate to operating profits, and it is estimated that no more than 1% of the optical fiber buried in recent years is actually 'lit'. While unused fiber may not be carrying traffic, it still has value as dark backbone fiber. Companies can lease or sell the unused fiber to other providers who are
looking for service in or through an area. Many companies are "overbuilding" their networks for the specific purpose of having a large network of dark fiber for sale. This is a great idea as many cities are difficult to deal with when applying for permits and trenching in new ducts is very costly. Modern cables come in a wide variety of sheathings and armor, designed for applications such as direct burial in trenches, dual use as power lines, installation in conduit, lashing to aerial telephone poles, submarine installation, or insertion in paved streets. In recent years the cost of small fiber-count pole-mounted cables has greatly decreased due to the high Japanese and South Korean demand for fiber to the home (FTTH) installations.
Optical fibers may be connected to each other by connectors or by splicing, that is , joining two fibers together to form a continuous optical waveguide. The generally accepted splicing method is arc fusion splicing, which melts the fiber ends together with an electric arc. For quicker fastening jobs, a "mechanical splice" is used. Fusion splicing is done with a specialized instrument that typically operates as follows: The two cable ends are fastened inside a splice enclosure that will protect the splices, and the fiber ends are stripped of their protective polymer coating (as well as the more sturdy outer jacket, if present). The ends are cleaved (cut) with a precision cleaver to make them perpendicular, and are placed into special holders in the splicer. The splice is usually inspected via a magnified viewing screen to check the cleaves before and after the splice. The splicer uses small motors to align the end faces together, and emits a small spark between electrodes at the gap to burn off dust and moisture. Then the splicer generates a larger spark that raises the temperature above the melting point of the glass, fusing the ends together permanently. The location and energy of the spark is carefully controlled so that the molten core and cladding don't mix, and this
minimizes optical loss. A splice loss estimate is measured by the splicer, by directing light through the cladding on one side and measuring the light leaking from the cladding on the other side. A splice loss under 0.1 dB is typical. The complexity of this process makes fiber splicing much more difficult than splicing copper wire. Mechanical fiber splices are designed to be quicker and easier to install, but there is still the need for stripping, careful cleaning and precision cleaving. The fiber ends are aligned and held together by a precision-made sleeve, often using a clear index-matching gel that enhances the transmission of light across the joint. Such joints typically have higher optical loss and are less robust than fusion splices, especially if the gel is used. All splicing techniques involve the use of an enclosure into which the splice is placed for protection afterward. Fibers are terminated in connectors so that the fiber end is held at the end face precisely and securely. A fiber-optic connector is basically a rigid cylindrical barrel surrounded by a sleeve that holds the barrel in its mating socket. The mating mechanism can be "push and click", "turn and latch" ("bayonet"), or screw-in (threaded). A typical connector is installed by preparing the fiber end and inserting it into the rear of the connector body. Quick-set adhesive is usually used so the fiber is held securely, and a strain relief is secured to the rear. Once the adhesive has set, the fiber's end is polished to a mirror finish. Various polish profiles are used, depending on the type of fiber and the application. For single-mode fiber, the fiber ends are typically polished with a slight curvature, such that when the connectors are mated the fibers touch only at their cores. This is known as a "physical contact" (PC) polish. The curved surface may be polished at an angle, to make an "angled physical contact" (APC) connection. Such connections have higher loss than PC connections, but greatly reduced back reflection, because light that reflects from the angled surface leaks out of the fiber core; the resulting loss in signal strength is known as gap loss. APC fiber ends have low back reflection even when disconnected.
OPTICAL FIRE CABLE TYPES:
In a loose-tube cable design, color-coded plastic buffer tubes house and protect optical fibers. A gel filling compound impedes water penetration. Excess fiber length (relative to buffer tube length) insulates fibers from stresses of installation and environmental loading. Buffer tubes are stranded around a dielectric or steel central member, which serves as an anti-buckling element. The cable core, typically uses aramid yarn, as the primary tensile strength member. The outer polyethylene jacket is extruded over the core. If armoring is required, a corrugated steel tape is formed around a single jacketed cable with an additional jacket extruded over the armor. Loose-tube cables typically are used for outside-plant installation in aerial, duct and directburied applications.
With tight-buffered cable designs, the buffering material is in direct contact with the fiber. This design is suited for "jumper cables" which connect outside plant cables to terminal equipment, and also for linking various devices in a premises network.
Multi-fiber, tight-buffered cables often are used for intra-building, risers, general building and plenum applications. The tight-buffered design provides a rugged cable structure to protect individual fibers during handling, routing and connectorization. Yarn strength members keep the tensile load away from the fiber. As with loose tube cables, optical specifications for tight-buffered cables also should include the maximum performance of all fibers over the operating temperature range and life of the cable. Averages should not be acceptable.
COLOUR CODE FOR OPTICAL FIBRE CABLE
Fiber optic cables are terminated using an industry standard color code. For cables that consist of more than 12 strands, the color code repeats itself. Each group of 12 strands is identified with some other means such as: Multiple buffer tubes each with 12 or less strands either numbered or colored following the same color code, e.g., 1st tube is blue, 2nd is orange, etc. 24 strand groups with the color code repeating with some variation, e.g., the 1st group of 12 strands are solid colors and the 2nd groups are solid colors with a stripe or some other identifying mark. The color sequence is illustrated below. It is very similar to the color code for twisted pair cables except the second group of colors is used first and 2 new colors are added at the end.
Fiber Number 1 2 3 4 5 6 7 8 9 10 11 12
Color Code blue orange green brown slate white red black yellow violet rose aqua
OPTICAL FIBRE CABLE SYSTEMS:
PDH: PLESIOCHRONOUS DIGITAL HIERARCHY
Plesiochronous Digital Hierarchy (PDH) is a transmission system for ’voice communication’ using plesiochronous synchronization. PDH can be (and is) used for data transmission. However, technology is optimized for transportation of PCM coded voice. PDH uses time division multiplexing (TDM) to control and to share resources of transmission line. Connections in PDH systems are identified by their respective positions in PDH frame . PDH E1 (basic rate PDH) uses 32 time−slot frame structure where each slot is 8 bits (i.e. one PCM coded voice sample). Frames are sent on every 125us (inter−sample time in PCM coding). 30 time−slots are used for information transfer and 2 for control purposes (i.e. TS 0 and TS 16). Therefore basic rate of PDH E1 is 8000Hz*8bits*32=2048 kbps. PDH hierarchy is based on multiplexing four lower level signals into one higher level signal. This is due to additional control information which is added on higher stages of multiplexing.
SDH: SYNCHRONOUS DIGITAL HIERARCHY
Synchronous Digital Hierarchy (SDH) is an international digital telecommunications network hierarchy which standardizes transmission from 155Mbps to 10Gbps. SDH is general purpose transport system which is not optimized to any particular purpose. However, it still carries dependencies to the old PCM and PDH world (i.e. Frames are generated on every 125us and
they are tiled to 8bit pieces). Frame in SDH is called as STM−1 (Synchronous Transport Module−Level 1). STM−1 frame size is 2430*8bits and it is best viewed as 9 times 270 matrix. First 9x9 bytes are header information which is used for control purposes. The SDH specifies how payload data is framed and transported synchronously across optical fiber transmission links without requiring all the links and nodes to have the same synchronized clock for data transmission and recovery (i.e. both the clock frequency and phase are allowed to have variations, or be plesiochronous). SDH offers several advantages over the PDH. Where PDH lacks built−in facilities for automatic management and routing, and locks users into proprietary methods, SDH can improve network reliability and performance, offers much greater flexibility and lower operating and maintenance costs, and provides for a faster provision of new services. Synchronization In digital telephone transmission, "synchronous" means the bits from one call are carried within one transmission frame. "Plesiochronous" almost (but not) synchronous," or a call that must be extracted from more than one transmission frame. In synchronization plan operator decides how timing information is spread all over the network. There are some timing requirements which needs to be met if overflowing of reception buffer are wanted to avoid (i.e. Frame slips where single frame is lost due to clock differences in transmitter and receiver).
WDM: WAVELENGTH DIVISON MULTIPLEXING
In fiber-optic communications, wavelength-division multiplexing (WDM) is a technology which multiplexes multiple optical carrier signals on a single optical fiber by using different wavelengths (colours) of laser light to carry different signals. This allows for a multiplication in
capacity, in addition to enabling bidirectional communications over one strand of fiber. This is a form of frequency division multiplexing (FDM) but is commonly called wavelength division multiplexing. The term wavelength-division multiplexing is commonly applied to an optical carrier (which is typically described by its wavelength), whereas frequency-division multiplexing typically applies to a radio carrier (which is more often described by frequency). However, since wavelength and frequency are inversely proportional, and since radio and light are both forms of electromagnetic radiation, the two terms are equivalent in this context.
A WDM system uses a multiplexer at the transmitter to join the signals together, and a demultiplexer at the receiver to split them apart. With the right type of fiber it is possible to have a device that does both simultaneously, and can function as an optical add-drop multiplexer. The optical filtering devices used have traditionally been etalons, stable solid-state single-frequency Fabry-Perot interferometers in the form of thin-film-coated optical glass. The concept was first published in 1970, and by 1978 WDM systems were being realized in the laboratory. The first WDM systems only combined two signals. Modern systems can handle up to 160 signals and can thus expand a basic 10 Gbit/s fiber system to a theoretical total capacity of over 1.6 Tbit/s over a single fiber pair. WDM systems are popular with telecommunications companies because they allow them to expand the capacity of the network without laying more fiber. By using WDM and optical amplifiers, they can accommodate several generations of technology development in their optical infrastructure without having to overhaul the backbone network. Capacity of a given link can be expanded by simply upgrading the multiplexers and demultiplexers at each end. This is often done by using optical-to-electrical-to-optical (O/E/O) translation at the very edge of the transport network, thus permitting interoperation with existing equipment with optical interfaces. Most WDM systems operate on single mode fiber optical cables, which have a core diameter of 9 µm. Certain forms of WDM can also be used in multi-mode fiber cables (also known as premises cables) which have core diameters of 50 or 62.5 µm. Early WDM systems were expensive and complicated to run. However, recent standardization and better understanding of the dynamics of WDM systems have made WDM less expensive to deploy. Optical receivers, in contrast to laser sources, tend to be wideband devices. Therefore the demultiplexer must provide the wavelength selectivity of the receiver in the WDM system.
CWDM in contrast to conventional WDM and DWDM uses increased channel spacing to
allow less sophisticated and thus cheaper transceiver designs. To again provide 16 channels on a single fiber CWDM uses the entire frequency band between second and third transmission window (1310/1550 nm respectively) including both windows (minimum dispersion window and minimum attenuation window) but also the critical area where OH scattering may occur, recommending the use of OH-free silica fibers in case the wavelengths between second and third transmission window shall also be used. WDM, DWDM and CWDM are based on the same concept of using multiple wavelengths of light on a single fiber, but differ in the spacing of the wavelengths, number of channels, and the ability to amplify the multiplexed signals in the optical space. EDFA provide an efficient wideband amplification for the C-band, Raman amplification adds a mechanism for amplification in the L-band. For CWDM wideband optical amplification is not available, limiting the optical spans to several tens of kilometres.
ADVANTAGES OF WDM:
The adoption of WDM allows, as a first step, to increase the transmission capacity of the backbones by reusing existing fibre cables. WDM combines cost-effectiveness with scalability.
The deployment of WDM offers the operator a number of advantages, but primarily, wholelife cost savings. WDM solutions can be deployed with one wavelength on day one, yet engineered with the capability for expansion to one hundred times that. By designing in scalability, network operators can lock onto savings as the network evolves to accommodate large capacity demands. Secondly optical components are inherently more reliable than their electronic counterparts, and network operators and end users clearly benefit from this. Thirdly, as new optical network elements become commercially available, these can be incorporated into the network, allowing a smooth evolution to an all-optical network. Recent technological advances (in RACE and ACTS) in WDM have moved key building blocks such as optical amplifiers and WDM transmitters from development to products.
Dense wavelength division multiplexing, or DWDM for short, refers originally to optical signals multiplexed within the 1550-nm band so as to leverage the capabilities (and cost) of erbium doped fiber amplifiers (EDFAs), which are effective for wavelengths between approximately 1525-1565 nm (C band), or 1570-1610 nm (L band). EDFAs were originally developed to replace SONET/SDH optical-electrical-optical (OEO) regenerators, which they have made practically obsolete. EDFAs can amplify any optical signal in their operating range, regardless of the modulated bit rate. In terms of multi-wavelength signals, so long as the EDFA has enough pump energy available to it, it can amplify as many optical signals as can be multiplexed into its amplification band (though signal densities are limited by choice of modulation format). EDFAs therefore allow a single-channel optical link to be upgraded in bit
rate by replacing only equipment at the ends of the link, while retaining the existing EDFA or series of EDFAs through a long haul route. Furthermore, single-wavelength links using EDFAs can similarly be upgraded to WDM links at reasonable cost. The EDFAs cost is thus leveraged across as many channels as can be multiplexed into the 1550-nm band. DWDM systems At this stage, a basic DWDM system contains several main components: A DWDM terminal multiplexer. The terminal multiplexer actually contains one wavelength converting transponder for each wavelength signal it will carry. The wavelength converting transponders receive the input optical signal (i.e., from a client-layer SONET/SDH or other signal), convert that signal into the electrical domain, and retransmit the signal using a 1550 nm band laser. (Early DWDM systems contained 4 or 8 wavelength converting transponders in the mid 1990s. By 2000 or so, commercial systems capable of carrying 128 signals were available.) The terminal mux also contains an optical multiplexer, which takes the various 1550 nm band signals and places them onto a single SMF-28 fiber. The terminal multiplexer may or may not also support a local EDFA for power amplification of the multiwavelength optical signal. An intermediate optical terminal, or Optical Add-drop multiplexer. This is a remote amplification site that amplifies the multi-wavelength signal that may have traversed up to 140 km or more before reaching the remote site. Optical diagnostics and telemetry are often extracted or inserted at such a site, to allow for localization of any fiber breaks or signal impairments. In more sophisticated systems (which are no longer point-to-point), several signals out of the multiwavelength signal may be removed and dropped locally.
A DWDM terminal demultiplexer. The terminal demultiplexer breaks the multi-wavelength signal back into individual signals and outputs them on separate fibers for client-layer systems (such as SONET/SDH) to detect. Originally, this demultiplexing was performed entirely passively, except for some telemetry, as most SONET systems can receive 1550-nm signals. However, in order to allow for transmission to remote client-layer systems (and to allow for digital domain signal integrity determination) such demultiplexed signals are usually sent to O/E/O output transponders prior to being relayed to their client-layer systems. Often, the functionality of output transponder has been integrated into that of input transponder, so that most commercial systems have transponders that support bi-directional interfaces on both their 1550-nm (i.e., internal) side, and external (i.e., client-facing) side. Transponders in some systems supporting 40 GHz nominal operation may also perform forward error correction (FEC) via 'digital wrapper' technology, as described in the standard.
Optical Supervisory Channel (OSC). This is an additional wavelength usually outside the
EDFA amplification band (at 1510 nm, 1620 nm, 1310 nm or another proprietary wavelength). The OSC carries information about the multi-wavelength optical signal as well as remote conditions at the optical terminal or EDFA site. It is also normally used for remote software upgrades and user (i.e., network operator) Network Management information. It is the multiwavelength analogue to SONET's DCC (or supervisory channel). ITU standards suggest that the OSC should utilize an OC-3 signal structure, though some vendors have opted to use 100 megabit Ethernet or another signal format. Unlike the 1550 nm band client signal-carrying
wavelengths, the OSC is always terminated at intermediate amplifier sites, where it receives local information before retransmission.
Module 5 NETWORKING
The Open systems Interconnection Reference Model (OSI Reference Model or OSI Model) is an abstract description for layered communications and computer network protocol design. It was developed as part of the Open Systems Interconnection (OSI) initiative. In its most basic form, it divides network architecture into seven layers which, from top to bottom, are the Application, Presentation, Session, Transport, Network, Data-Link, and Physical Layers. It is therefore often referred to as the OSI Seven Layer Model. A layer is a collection of conceptually similar functions that provide services to the layer above it and receives service from the layer below it.
Application (Layer 7)
This layer supports application and end-user processes. Communication partners are identified, quality of service is identified, user authentication and privacy are considered, and any constraints on data syntax are identified. Everything at this layer is application-specific. This layer provides application services for file transfers, e-mail, and other network software services. Telnet and FTP are applications that exist entirely in the application level. Tiered application architectures are part of this layer.
Presentation (Layer 6)
This layer provides independence from differences in data representation (e.g., encryption) by translating from application to network format, and vice versa. The presentation layer works to transform data into the form that the application layer can accept. This layer formats and encrypts
data to be sent across a network, providing freedom from compatibility problems. It is sometimes called the syntax layer. This layer establishes, manages and terminates connections between applications. The session layer sets up, coordinates, and terminates conversations, exchanges, and dialogues between the applications at each end. It deals with session and connection coordination. This layer provides transparent transfer of data between end systems, or hosts, and is responsible for end-to-end error recovery and flow control. It ensures complete data transfer. This layer provides switching and routing technologies, creating logical paths, known as virtual circuits, for transmitting data from node to node. Routing and forwarding are functions of this layer, as well as addressing, internetworking, error handling, congestion control and packet sequencing. At this layer, data packets are encoded and decoded into bits. It furnishes transmission protocol knowledge and management and handles errors in the physical layer, flow control and frame synchronization. The data link layer is divided into two sub layers: The Media Access Control (MAC) layer and the Logical Link Control (LLC) layer. The MAC sub layer controls how a computer on the network gains access to the data and permission to transmit it. The LLC layer controls frame synchronization, flow control and error checking. This layer conveys the bit stream - electrical impulse, light or radio signal -- through the network at the electrical and mechanical level. It provides the hardware means of sending and receiving data on a carrier, including defining cables, cards and physical aspects. Fast Ethernet, RS232, and ATM are protocols with physical layer components.
Session (Layer 5) Transport (Layer 4) Network (Layer 3)
Data Link (Layer 2)
Physical (Layer 1)
How Data Flows
TCP/IP is a suite of protocols. The acronym TCP/IP means "Transmission Control Protocol/Internet Protocol". It comes from the names of the two major protocols in the suite of protocols, i.e. the TCP and IP protocols).
In some ways, TCP/IP represents all communication rules for the internet and is based on the IP addressing notion, i.e. the idea of providing an IP address for each machine on the network so as to be able to route data packets. Given that the TCP/IP protocol suite was originally created with a military purpose, it is designed to respond to a certain number of criteria, including:
• • • •
Splitting messages into packets. Use of an address system. Routing data over the network. Error detection in data transmissions.
During a transmission, data crosses each one of the layers at the level of the originator machine. At each layer, a piece of information is added to the data packet, this is a header, a collection of information which guarantees transmission. At the level of the recipient machine, when passing through each layer, the header is read, and then deleted. So, upon its receipt, the message is in its original state...
At each level, the data packet changes aspect, because a header is added to it, so the designations change according to the layers:
The data packet is called a message at Application layer level The message is then encapsulated in the form of a segment in the Transport layer
Once the segment is encapsulated in the Internet layer it takes the name of datagram Finally, we talk about a frame at the Network Access layer level
The Data Encapsulation Process 1. One computer requests to send data to another over a network. 2. The data message flows through the Application Layer by using a TCP or UDP port to pass onto the internet layer. 3. The data segment obtains logical addressing at the Internet Layer via the IP protocol, and the data is then encapsulated into a datagram. 4. The datagram enters the Network Access Layer, where software will interface with the physical network. A data frame encapsulates the datagram for entry onto the physical network. At the end
of the process, the frame is converted to a stream of bits that is then transmitted to the receiving computer. 5. The receiving computer removes the frame, and passes the packet onto the Internet Layer. The Internet Layer will then remove the header information and send the data to the Transport layer. Likewise, the Transport layer removes header information and passes data to the final layer. At this final layer the data is whole again, and can be read by the receiving computer if no errors are present. The TCP/IP model, inspired by the OSI model, also uses the modular approach (use of modules or layers) but only contains four:
The roles of the different layers are as follows: • Network Access layer: specifies the form in which data must be routed whichever type of network is used. • Internet layer: responsible for supplying the data packet (datagram) • Transport layer: provides the routing data, along with the mechanisms making it possible to know the status of the transmission • Application layer: incorporates standard network applications (Telnet, SMTP, FTP,).
Network Access layer
The network access layer is the first layer of the TCP/IP stack; it offers the ability to access whichever physical network, i.e. the resources to be implemented so as to transmit data via a network. So, the network access layer contains all specifications relating to the transmission of data over a physical network, when it is a local area network (Token ring, Ethernet, FDDI), connected by telephone line or any other type of link to a network. It deals with the following concepts:
• • • • •
Routing data over the connection Coordination of the data transmission (synchronisation) Data format Signal conversion (analogue/digital) Error detection on arrival.
The Internet layer
The Internet layer is the "most important" layer (they are all important in their way) because it is this which defines the datagram and manages the IP addressing notions. It enables the routing of datagram (data packets) to remote machines along with the management of their division and assembly upon receipt.
The Transport layer
The protocols for the preceding layers make it possible to send information from one machine to another. The transport layer enables applications running on remote machines to communicate. The problem is identifying these applications. In fact, depending on the machine and its operating system, the application may be a program, task, process... Furthermore, the name of the application may vary from system to system, that is why a numbering system has been put in place so as to be able to associate an application type with a data type, these identifiers are called ports. The transport layer contains two protocols enabling two applications to exchange data independently of the type of network taken (i.e. independently of the lower layers), these are the following two protocols:
TCP, a connection orientated protocol which provides error detection. UDP, a connectionless orientated protocol where error detection is outdated.
The Application layer
The application layer is located at the top of the TCP/IP protocol layers. This one contains the network applications which make it possible to communicate using the lower layers. The software in this layer therefore communicates using one of the two protocols of the layer below (the transport layer), i.e. TCP or UDP. There are different types of applications for this layer, but the majorities are network services, or applications supplied to the user to provide the interface with the operating system. They can be classed according to the services that they provide: File and print management services (transfer) Network connection services Remote connection services Various Internet utilities
• • • o
IP protocol is part of the Internet layer of the TCP/IP protocol suite. It is one of the most important Internet protocols because it allows the development and transport of IP datagram’s (data packets), without however ensuring their "delivery". In reality, IP protocol processes IP datagram’s independently from each other by defining their representation, routing and forwarding. IP protocol determines the recipient of the message using 3 fields:
• • •
The IP address field: machine address The subnet mask field: a subnet mask enables the IP protocol to establish the part of the IP address which relates to the network The default gateway field: enables the Internet protocol to know which machine to deliver a datagram to if ever the destination machine is not on the local area network.
The aim of TCP
Using the TCP protocol, applications can communicate securely (thanks to the TCP protocol's acknowledgements system), independently from the lower layers. This means that routers (which work in the internet layer) only have to route data in the form of datagram, without being concerned with data monitoring because this is performed by the transport layer (or more specifically by the TCP protocol). During a communication using the TCP protocol, the two machines must establish a connection. The originator machine (the one which requests the connection) is called the client, while the recipient machine is called the server. So it is said that we are in a Client-Server environment. The machines in such an environment communicate in online mode, i.e. the communication takes place in both directions. To enable the communication and all the controls which accompany it to operate well, the data is encapsulated, i.e. a header is added to data packets which will enable the transmissions to be synchronised and ensure their reception. Another feature of TCP is the ability to control the data speed using its capability to issue variably sized messages, these messages are called segments.
The multiplexing function
TCP makes it possible to carry out an important task: multiplexing/demultiplexing, i.e. to convey data from various applications on the same line or in other words put information arriving in parallel into order.
These operations are conducted using the concept of ports (or sockets), i.e. a number linked to an application type which, when combined with an IP address, makes it possible to uniquely determine an application which is running on a given machine.
Use of an address system
Computers communicate over the Internet using the IP protocol (Internet Protocol), which uses numerical addresses, called IP addresses, made up of four whole numbers (4 bytes) between 0 and 255 and written in the format xxx.xxx.xxx.xxx. For example, 22.214.171.124 is an IP address given in technical format.
These addresses are used by networked computers to communicate, so each computer on a network has a unique IP address on that network. Every network interface on a TCP/IP device is identified by a globally unique IP address. Host devices, for example, PCs, typically have a single IP address. Routers typically have two or more IP addresses, depending on the number of interfaces they have. Each IP address is 32 bits long and is composed of four 8-bit fields called octets. This address is normally represented in “dotted decimal notation” by grouping the four octets and representing each octet in decimal form. Each octet represents a decimal number in the range 0-255. For example, 11000001 10100000 00000001 00000101, is known as 126.96.36.199. Each IP address defines the network ID and host ID of the device. The network ID part of the IP address is centrally administered by the Internet Network Information Centre (Inter NIC) and is unique throughout the Internet. The host ID is assigned by the authority which controls the network. The network ID identifies the systems that are located on the same network or subnet. The network ID must be unique to the internetwork. The host ID identifies a TCP/IP network device (or host) within a network. The address for each host must be unique to the network ID. In the example above, the PC is connected to network “188.8.131.52” and has a unique host ID of “.5”.
When the host-id is cancelled, i.e. when the bits reserved for the machines on the network are replaced by zeros (for example 184.108.40.206), something called a network address is obtained. This address cannot be allocated to any of the computers on the network.
When the net ID is cancelled, i.e. when the bits reserved for the network are replaced by zeros, a machine address is obtained. This address represents the machine specified by the host-ID which is found on the current network.
When all the bits of the host-id are at 1, the address obtained is called the broadcast address. This a specific address, enabling a message to be sent to all the machines on the network specified by the net ID Conversely, when all the bits of the net ID are at 1, the address obtained is called the multicast address. Finally the address 127.0.0.1 is called the loopback address because it indicates the local host.
IP addresses are divided into classes, according to the number of bytes which represent the network.
In a class A IP address, the first byte represents the network. The most significant bit (the first bit, that to the left) is at zero which means that there are 27 (00000000 to 01111111) network possibilities, which is 128 possibilities However, the 0 network (bits valuing 00000000) does not exist and number 127 is reserved to indicate your machine. The networks available in class A are therefore networks going from 220.127.116.11 to 18.104.22.168 (the last bytes are zeros which indicate that this is indeed a network and not computers!) The three bytes to the left represent the computers on the network, the network can therefore contain a number of computers equal to: 224-2 = 16,777,214 computers. A class A IP address, in binary looks like: 0 xxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx Network Computers
In a class B IP address, the first two bytes represent the network. The first two bits are 1 and 0, which means that there are 214 (10 000000 00000000 to 10 111111 11111111) network possibilities, which is 16,384 possible networks. The networks available in class B are therefore networks going from 22.214.171.124 to 126.96.36.199.
The two bytes to the left represent the computers on the network. The network can therefore contain a number of computers equal to: 216-21 = 65,534 computers.
A class B IP address, in binary looks like: 10 xxxxxx xxxxxxxx xxxxxxxx xxxxxxxx Network Computers
In a class C IP address, the first three bytes represent the network. The first three bits are 11 and 0 which means that there are 221 network possibilities, i.e. 2,097,152. The networks available in class C are therefore networks going from 192.0.0.0 to 188.8.131.52. The byte to the left represents the computers on the network, the network can therefore contain: 28-21 = 254 computers. In binary, a class C IP address looks like: 110 xxxxx xxxxxxxx xxxxxxxx xxxxxxxx Network Computers
Allocation of IP addresses
The aim of dividing IP addresses into three classes A, B and C is to make the search for a computer on the network easier. In fact, with this notation it is possible to firstly search for the network that you want to reach, then search for a computer on this network. So, allocation of IP address is done according to the size of the network. Class Number of possible networks Maximum number of computers on each one A B C 126 16384 2097152 16777214 65534 254
Class A addresses are used for very large networks, while class C addresses are for example allocated to small company networks.
Reserved IP addresses
It frequently happens that in a company or organization only one computer is linked to the Internet and it is through this that other computers on the network access the Internet (generally we talk of a proxy or gateway). In such a case, only the computer linked to the network needs to reserve an IP address with ICANN. However, the other computers still need an IP address to be able to communicate with each other internally. So, ICANN has reserved a handful of addresses in each class to enable an IP address to be allocated to computers on a local network linked to the Internet without the risk of creating IP address conflicts on the network of networks. These are the following addresses: Private class A IP addresses: 10.0.0.1 to 10.255.255.254, enabling the creation of large private networks comprising of thousands of computers. • Private class B IP addresses: 172.16.0.1 to 172.31.255.254, making it possible to create medium sized private networks. • Private class C IP addresses: 192.168.0.1 to 192.168.0.254, for putting in place small private networks.
In short, a mask is produced containing 1s with the location of bits that you want to keep and 0s for those you want to cancel. Once this mask is created, you simply put a logical AND between the values you want to mask and the mask in order to keep the part you wish to cancel separate from the rest. So a net mask is presented in the form of 4 bytes separated by dots (like an IP address), it comprises (in its binary notation) zeros at the level of the bits from the IP address that you wish to cancel (and ones at the level of those you want to keep).
Importance of subnet masks
The primary importance of a subnet mask is to enable the simple identification of the network associated to an IP address. Indeed, the network is determined by a certain number of bytes in the IP address (1 byte for class A addresses, 2 for class B and 3 bytes for class C). However, a network is written by taking the number of bytes which characterize it, then completing it with zeros. For example, the network linked to the address 184.108.40.206 is 220.127.116.11, because it is a class A type IP address. To find out the network address linked to the IP address 18.104.22.168, you simply need to apply a mask where the first byte is only made up of 1s (which is 255 in decimal), then 0s in the following bytes. The mask is: 11111111.00000000.00000000.00000000
the mask associated with the IP address 22.214.171.124 is therefore 255.0.0.0. The binary value of 126.96.36.199 is: 00100010.11010000.01111011.00001100 so an AND logic between the IP address and the mask gives the following result:
00100010.11010000.01111011.00001100 AND 11111111.00000000.00000000.00000000 = 00100010.00000000.00000000.00000000 Which is 188.8.131.52? It is the network linked to the address 184.108.40.206 By generalizing, it is possible to obtain masks relating to each class of address: For a Class A address, only the first byte must be retained. The mask has the following format 11111111.00000000.00000000.00000000, i.e. 255.0.0.0 in decimal; • For a Class B address, the first two bytes must be retained, which gives the following mask 11111111.11111111.00000000.00000000, relating to 255.255.0.0 in decimal; • For a Class C address, by the same reasoning, the mask will have the following format 11111111.11111111.11111111.00000000, i.e. 255.255.255.0 in decimal;
Creation of subnets
Let us re-examine the example of the network 220.127.116.11, and assume that we want the first two bits of the second byte to make it possible to indicate the network. The mask to be applied will then be: 11111111.11000000.00000000.00000000 That is 255.192.0.0 If we apply this mask to the address 18.104.22.168 we get: 22.214.171.124 In reality there are 4 possible scenarios for the result of the masking of an IP address of a computer on the network 126.96.36.199 When the first two bits of the second byte are 00, in which case the result of the masking is 188.8.131.52 • When the first two bits of the second byte are 01, in which case the result of the masking is 184.108.40.206 • When the first two bits of the second byte are 10, in which case the result of the masking is 220.127.116.11 • When the first two bits of the second byte are 11, in which case the result of the masking is 18.104.22.168
Therefore, this masking divides a class A network (able to allow 16,777,214 computers) into 4 subnets - from where the name of subnet mask - can allow 222 computers or 4,194,304 computers.
It may be interesting to note that in these two cases, the total number of computers is the same, which is 16,777,214 computers (4 x 4,194,304 - 2 = 16,777,214). The number of subnets depends on the number of additional bits allocated to the network (here 2). The number of subnets is therefore:
Routing data over the network.
Routers are devices which make it possible to "choose" the path that datagrams will take to arrive at the destination. They are machines with several network interface cards each one of which is linked to a different network. So, in the simplest configuration, the router only has to "look at" what network a computer is located on to send datagrams to it from the originator. However, on the Internet the schema is much more complicated for the following reasons: The number of networks to which a router is connected is generally large The networks to which the router is linked can be linked to other networks that the router cannot see directly
So, routers work using routing tables and protocols, according to the following model: The router receives a frame from a machine connected to one of the networks it is attached to • Datagrams are sent on the IP layer. • The router looks at the datagram's header • If the destination IP address belongs to one of the networks to which one of the router interfaces is attached, the information must be sent at layer 4 after the IP header has been un encapsulated (removed) • If the destination IP address is part of a different network, the router consults its routing table, a table which establishes the path to take for a given address. • The router sends the datagram using the network interface card linked to the network on which the router decides to send the packet.
So, there are two scenarios, either the originator and recipient belong to the same network in which case we talk about direct delivery, or there is at least one router between the originator and recipient, in which case we talk about indirect delivery. In the case of indirect delivery, the role of the router and in particular that of the routing table is very important. So, the operation of a router is determined by the way in which this routing table is created. If the routing table is entered manually by the administrator, it is a static routing (suitable for small networks) • If the router builds its own routing tables using information that it receives (via the routing protocols), it is a dynamic routing.
Routes to destinations are set up manually. Network reachability is not dependent on existence and state of network. Route may be up or down but static routes will remain in routing tables and traffic would still be sent towards route. Not suitable for large number of network. It is also known as non-adaptive routing.
Routes are made via internal and external routing protocols. Network reachability is dependent on the existence and State of network. It is also known as adaptive routing. The routing protocols are: - RIP (routing information protocols)
- OSPF (open shortest path first)
A sub network, or subnet, describes networked computers and devices that have a common, designated IP address routing prefix. Subnetting is used to break the network into smaller more efficient subnets to prevent excessive rates of Ethernet packet collision in a large network. Such subnets can be arranged hierarchically, with the organization's network address space (see also Autonomous System) partitioned into a tree-like structure. Routers are used to manage traffic and constitute borders between subnets. A routing prefix is the sequence of leading bits of an IP address that precede the portion of the address used as host identifier. In IPv4 networks, the routing prefix is often expressed as a "subnet mask", which is a bit mask covering the number of bits used in the prefix. An IPv4 subnet mask is frequently expressed in quad-dotted decimal representation, e.g., 255.255.255.0 is the subnet mask for the 192.168.1.0 network with a 24-bit routing prefix (192.168.1.0/24). All hosts within a subnet can be reached in one "hop" (time to live = 1), implying that all hosts in a subnet are connected to the same link. A typical subnet is a physical network served by one router, for instance an Ethernet network (consisting of one or several Ethernet segments or local area networks, interconnected by network switches and network bridges) or a Virtual Local Area Network (VLAN). However, subnetting allows the network to be logically divided regardless of the physical layout of a network, since it is possible to divide a physical network into several subnets by configuring different host computers to use different routers.
While improving network performance, subnetting increases routing complexity, since each locally connected subnet is typically represented by one row in the routing tables in each connected router. However, with a clever design of the network, routes to collections of more distant subnets within the branches of a tree-hierarchy can be aggregated by single routes. Existing subnetting functionality in routers made the introduction of Classless Inter-Domain Routing seamless.
•Class A natural mask 255.0.0.0 •Class B natural mask 255.255.0.0 •Class C natural mask 255.255.255.0 By separating the network and host IDs of an IP address, masks facilitate the creation of subnets. With the use of masks, networks can be divided into sub networks by extending the network IDs of the address into the host ID. Subnetting increases the number of sub networks and reduces the number of hosts. Defining a subnet mask based on the number of subnets required • Add two to the number of subnets required and convert to binary • Count the number of bits required • Convert the required number of bits to decimal in high order • Example: Class C address, 5 subnets required 7 converted to binary is 110 ( 3 bits) Three bits are required so configure the first three bits of the host ID as the subnet ID The decimal value for 1110 0000 is 224 The subnet mask is 255.255.255.224 for this class C address
If you are dividing your network into subnets, you need to define a subnet mask. Follow these steps:
1. Determine the number of subnets you require. Add two to the number of subnets required and convert to binary.
2. Count the number of bits required to represent the number of physical segments in binary. For example, if you need five subnets, the binary value of seven is 110. Representing seven in binary requires three bits. 3. Convert the required number of bits to decimal format in high order (from left to right). For example, if three bits are required, configure the first three bits of the host ID as the subnet ID. The decimal value for binary 11100000 is 224. The subnet mask is 255.255.225.224 (for a Class C address).
Defining a subnet mask based on the number of hosts
• • • • • Add two to the number of hosts required and convert the sum to binary Count the number of bits required for the host portion Subtract this number from the total number of bits in the host ID Convert the required number of bits to decimal in high order Example: Class B address, 2000 devices per subnet required 2002 converted to binary is 11111010010 ( 11 bits) Eleven bits are required for the host so configure the first five bits of the host ID as the subnet ID (16 - 11 = 5) The decimal value for 1111 1000 is 248 The subnet mask is 255.255.248.0 for a class B address If you do not want all your hosts to be on the same subnet, you need to define a subnet mask, assuming that you have been allocated a single network address. Follow these steps: 1. Decide on the number of hosts you want to have on each subnet. Convert this number to binary format. 2. Count the number of bits required to represent the number of hosts in binary. For example, if you want up to 2,000 hosts per subnet, the binary value for 2002 is 11111010010. Representing 2,002 in binary requires 11 bits. To calculate the number of bits required for the mask, subtract the number of bits required for the host from the total number of bits in the host. In this example the result is five (16 - 11). 4. Convert the required number of bits to decimal format in high order (from left to right). In this example, five bits are required. Configure the first five bits of the host ID as the subnet ID. The decimal value for 11111000 is 248. The subnet mask is 255.255.248.0 (for a class B address). The subnet conversion table above shows all the possible combinations of subnets and hosts for a Class C network address. For example, if we want to implement five subnets, we would use a subnet mask of 255.255.255.224. This would allow up to a maximum of six subnets with 30 devices per subnet. If there are zero bits in the subnet
mask we are not using subnetting and are left with the default of one network with 254 hosts. We cannot just use one bit in the subnet mask because the only subnet IDs would be 0 and 1
neither of which are valid. Similarly we cannot use 7 bits in the subnet ID because the only host IDs would again be 0. The subnet conversion table above shows all the possible combinations of subnets and hosts for a Class C network address. For example, if we want to implement five subnets, we would use a subnet mask of 255.255.255.224. This would allow up to a maximum of six subnets with
30 devices per subnet. If there are zero bits in the subnet mask we are not using subnetting and are left with the default of one network with 254 hosts. We cannot just use one bit in the subnet mask because the only subnet IDs would be 0 and 1 neither of which are valid. Similarly we cannot use 7 bits in the subnet ID because the only host IDs would again be 0. When a portion of the address, blocked out by the subnet mask changes, the network devices know that these addresses are in different subnets. For example, for all addresses between 16 and 31 in the diagram above, the 4 bits blocked by the mask are 0001. These are on the same subnet. Therefore, for address 32 which is binary 0010 0000, we can see that the four bits blocked by the mask portion have changed. Therefore this must be a different subnet. Note: in the example above, 16 is the subnet ID but it is not a valid host ID since 16 = 0001 0000 and we cannot have all zeros in the host portion. Similarly 31 is not a valid host ID since 31 = 0001 1111 which is the broadcast address for this subnet. Subnet IDs comprised of all 0s or all 1s are called special case subnet addresses. A subnet ID of all 1s indicates a subnet broadcast while a subnet ID of all 0s indicates “this subnet”. When subnetting it is strongly recommended not to use these subnet IDs. However, it is possible to use these special case subnet addresses if they are supported by all routers and hardware on the network. Request For Comment (RFC) 950 details the limitations imposed when using special case addresses.
Shortcut method for defining Subnet ID’s using the Subnet Conversion Table
• From the maximum number of hosts Add 2 to the maximum number of hosts and this gives the first valid subnet ID. All subsequent IDs are multiples of the first valid subnet ID. Example: maximum number of hosts = 14 14+2=16 Sub net IDs = 16, 32, 48, 64,…….. • From the maximum number of subnets Add 2 to the maximum number of subnets. Divide 256 by this number and the result is the first valid subnet ID. All subsequent ID’s are multiples of the first valid subnet ID. Example: maximum number of subnets = 14 14+2=16 256/16 = 16 Sub net Ids = 16, 32, 48, 64.
There are two shortcut methods to define the subnet ID 1. Based on the subnet conversion table. This is described in the overhead above. 2. Based on the number of bits in the host portion. This is described in the following text. Shortcut method for defining subnet IDs from the number of bits in the host portion. Count the number of bits in the host ID portion. Multiply this number by a power of two and this is the first valid subnet ID. All subsequent subnet IDs are multiples of the first valid subnet ID. Mask = 255.255.255.192 192 = 1100 0000 Six bits in host portion 2^6=64 Subnet IDs 0, 64, 128, 192 Mask = 255.255.255.224 224 = 1110 0000 Five bits in host portion 2^5=32 Subnet IDs 0, 32, 64, 96, 128, 160, 192, 224 Mask = 255.255.255.240
240 = 1111 0000 Four bits in host portion 2^4=16 Subnet IDs = 0,16, 32, 48, 64, 80, 96, 112, 128, 144, 160, 176, 192, 208, 224, 240
Mask = 255.255.255.248. 248 = 1111 1000 Three bits in host portion 2^3=8 Subnet IDs = 0, 8, 16, 24, 32, 40,……………., 224, 232, 240, 248
Mask = 255.255.255.252
252 = 1111 1100 Two bits in host portion 2^2=4 Subnet IDs = 0, 4, 8, 12, 61, 20,…………., 240, 244, 248, 252 Note; in the last example there are only two valid host IDs on each subnet. For example; in subnet ID = 4 address 5 and 6 are the only two valid source addresses
the example above, a small company has been assigned a single Class C network. Without subnetting, up to a maximum of 254 hosts can share this network. In this configuration, if one device sends out an IP broadcast (e.g. DHCP Discover message) it will be received by every device on the network. To improve performance, the network administrator may reduce the number of devices that receive the broadcast by splitting the network into smaller subnets separated by a router. In the example above, the network has been split into six smaller subnets with a maximum of 30 hosts on each subnet. Note: the total maximum number of hosts on the network has been reduced from 254 to 180 hosts. Consult the subnet conversion table for all possible combinations of hosts and subnets.
The subnet conversion table above shows all the possible combinations of subnets and hosts on a class B network address. For example, if we want to implement subnets with approximately 100 devices on each we would use a subnet mask of 255.255.255.128. This would allow up to a maximum of 510 subnets with 126 devices on each. A commonly used subnet mask in class B networks is 255.255.255.0. This allows for 254 subnets with 254 devices each.
Static Subnetting vs. Variable Length Subnetting
Static subnetting means that all subnets in the subnetted network use the same subnet mask Simple to implement and easy to maintain, but results in wasted address space for small networks.
o For example, a network of four hosts that uses a subnet mask of 255.255.255.0 wastes 250 IP addresses
Variable Length subnetting implies that the sub networks that make up the network may use different subnet masks • A small subnet with only a few hosts needs a subnet mask that accommodates only these few hosts Each host on a TCP/IP network requires a subnet mask. A default subnet mask is used when a network is not divided into subnets. A customized subnet mask is used when a network is divided into subnets. In a default subnet mask, all bits that correspond to the network ID are set to 1. The decimal value in each of these octets is 255. All bits that correspond to the host ID are set to 0. For example, the class B address 22.214.171.124 has a network ID of 126.96.36.199 and a host ID 100.10. The default mask is therefore 255.255.0.0. There are two types of subnetting: static and variable length.
Variable Length Subnet Mask (VLSM)
• Variable Length Subnet Mask (VLSM) refers to the fact that one subnet network can be configured with different masks • 252 (1111 1100) - 62 subnets with 2 hosts each • 248 (1111 1000) - 30 subnets with 6 hosts each. • 240 (1111 0000) - 14 subnets with 14 hosts each. • 224 (1110 0000) - 6 subnets with 30 hosts each. • 192 (1100 0000) - 2 subnets with 62 hosts each.
Variable Length Subnet Mask (VLSM) refers to the fact that one network can be configured with different masks. The idea behind Variable Length Subnet Masks is to offer more flexibility in dividing a network into multiple subnets while still maintaining an adequate number of hosts in each subnet. Without VLSM, one subnet mask only can be applied to a network. This restricts the number of hosts given the number of subnets required. If you pick the mask so that you have enough subnets, you might not be able to allocate enough hosts in each subnet. The same is true for the hosts; a mask that allows enough hosts might not provide enough subnet space. Suppose for example, you were assigned a Class C network 188.8.131.52 and you need to divide that network into three subnets, with 50 hosts in one subnet and 25 hosts for each of the remaining subnets. Without subnetting you have 254 addressees available, 184.108.40.206 to 220.127.116.11. The desired subdivision cannot be done without VLSM, as we shall see. There are a handful of subnet masks of the form 255.255.255.X that can be used to divide the class C network 18.104.22.168 into more subnets. Remember that a mask should have a contiguous number of one starting from the left (network portion) and the rest of the bits should be zeros. The masks shown in the diagram above could be used to segment the 254 addresses available to you into more subnets.
The physical topology of a network refers to the configuration of cables, computers, and other peripherals. The way in which the network is laid physically or logically.
Every computer is connected to each other. Advantages: • • • • Dedicated link eliminates traffic problem. Failure of link does not affect other link. Privacy of link is maintained. Fault identification and fault isolation is easy.
Disadvantages: • • Hardware requirements and i/o ports increases. More space is required.
Each computer has a dedicated link to a central Controller called HUB or SWITCH. It acts as an exchange. Advantages:
• • •
Less expensive. Fault identification and isolation is easy. Failure of one link does not affect other.
Disadvantages: • If HUB or SWITCH fails, then whole network will be down.
There is one main backbone (central) HUB and secondary HUB emerges out from it. Central HUB is active while secondary can be active or passive. Active HUB contains Repeater. Advantages: • More and more devices can be attached. • Allows network to identify and isolate faults. • Point to point wiring for individual segments. Disadvantages: • If backbone gets down then whole system fails. • Difficult to configure.
It consists of a main run of cable with a terminator at each end.
Advantages: • • Ease in installation. Uses less cable.
Disadvantages: • • If backbone fails, whole network ceases. Difficult to add new devices.
Each device is connected to only two devices. Advantages: • Easy in installation. • To add and delete only 2 connections are altered. • Fault isolation is simplified. Disadvantages: • Break in ring ceases whole network. • Difficult to add new devices.
Combination of all the topologies. Advantages: Increased number of devices. Contains all topologies
Search engine: - www.google.com Books: - As par given by the institute Internet sites: - www.123eng.com, www.compnetworking.about.com
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