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For circuit-switched communication, the network sets up a connection to a UE/MS by

providing a fixed bandwidth allocation on the air interface. Even if only small amounts

of data are transferred, the UE/MS occupies the radio resource for the duration of the

connection. The user must pay for the total connection time.

Due to the constant bandwidth allocation, delays are minimized and the Quality of

Service (QoS) perception for real time services is very good.

Circuit-switched communication is suitable for data traffic when one or more of the

following cases apply:

1. Constant bandwidth data flow

2. Data is sensitive to even small connection delays

For example circuit-switched communication could be chosen for videoconferences

because of its sensitivity to connection delays. The video conferencing in the

implementation of 3G will be done using Circuit switched Data services.

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For packet-switched communication, the network delivers data packets as the need

arises. On the air interface, radio channels are shared as an access resource, between

severalUEs/MSssimultaneously. For WCDMA Systems, if the data rate for a given

user connection exceeds a certain threshold, the UE may temporarily be assigned a

dedicated resource on the air interface. The UE will drop back to a shared resource

when the source data rate is reduced.

Address information is included with each packet to enable the packet to find its

addressee. Packet-switched communication is suitable for data traffic when one or

more of the following cases apply:

1. Data is sent in bursts

2. Data is sensitive to errors

For example packet-switched communication should be chosen for telemetry

applications and e-mail, the former because of its sensitivity to errors and the latter

because the data is sent in bursts.

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Standard Bodies

All computer data networks were proprietary in the early 1970s. That is, the same

manufacturer, for example, IBM, provided the custom computer hardware and

software on each end of a communications link.

Further more, these proprietary networks were host centric. This meant that if your

mainframe host computer was IBM, for example, the networking front end

processors also had to be IBM. It also meant that data transmitted from any one of

your sites was first sent to the mainframe host computer for routing to another site.

The networking protocols and applications programs were proprietary and would not

inter-operate between manufacturers.

LAN technologies Token Ring and Ethernet would not inter-operate. Furthermore,

neither Token Ring nor Ethernet would inter-operate with a non-proprietary WANprotocol. Each manufacturer had their own proprietary communication protocols and

application programs, which had common code to tie them together.


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The International Standards Organization (ISO) has produced a protocol standard known as the OpenSystems Interconnection (OSI) Reference Model. This consists of 7-layers that describe the hierarchicaloperation of specific functions in the communications process. Although the protocol itself has notgained wide acceptance, it is considered important as a means of identifying the factors andcomparing the performance and capabilities of different protocols.

Each layer performs a well defined function, exchanging messages (relating to user data and controlinformation) with the equivalent layer in another system, and having a well-defined interface to thelayers immediately above and below itself.

OSI Reference Model - Layers

Characteristics of the OSI Layers

The OSI model is a modular design. Each successive layer of the OSI model works with the one aboveand below it.

The first four layers, Physical, Data Link, Network and Transport Layers, provide the end-to-endservices necessary for the transfer of data between two systems. These layers provide the protocolsassociated with the communications network used to link, the two computers together.

The top three layers, the Application, Presentation and Session Layers, provide the application servicesrequired for the exchange of information. That is, they allow two applications, each running on a

different node of the network, to interact with each other through the services provided by theirrespective operating systems.

The upper layers of the OSI model deal with application issues and generally are implemented only insoftware. The highest layer, application, is closest to the end user. Both users and application-layerprocesses interact with software applications that contain a communications component.

Reminders

Several phases are popularly used to help people remember the order of the layers of the OSI model.

From top to bottom the phrase, AllPeople Seem To Need Data Processing,may be remembered,referring to Application, Presentation, Session, Transport, Network, Datalink and Physical.

Alternatively, from bottom to top the phrase, Please Do Not Throw Sausage Pizza Away, may beremembered, referring to Physical, Datalink, Network, Transport, Session, Presentation andApplication.


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Physical Layer

The functions of the Physical Layer are to specify physical network structures,mechanical and electrical specifications for the transmission medium, and encodingand timing rules for bit transmission.

Specifications

These specifications can be grouped into a number of different areas, which includethe following:

Connection types

Physical topology

Signalling

Bit synchronisation

Bandwidth use

Multiplexing.

Connections

Networks use point-to-point and multipoint connections. The two connection typesdiffer in the manner in which devices connect to a cable or segment of transmissionmedium.


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Data Link LayerFunctions

The functions of the Data Link Layer are to: organise the physical layersbits into logical groups of information called frames,detect, and sometimes correct errors, control the flow of data and identify computers on the network with the use of mediaaccess control or MAC addresses.

Note: The Data Link Layer functions are usually divided into Media Access Control (MAC) functions and the Logical Link Control(LLC) functions.

MAC Data Link Sub-Layer

The MAC data link layer is concerned with the following:

Logical topology

Media access

MAC addressing.

Logical Topology

The actual signal path that data takes on a network is called the networkslogical topology. In a logical bus topology, every signalis received by all devices. In a logical ring topology, each device only receives signals that have been specifically sent to it.

Sometimes, the physical topology of a network does not reflect its logical topology. For example, in an IBM token ring network,the physical topology is a star, while the logical topology is a ring.

Media Access

Logical topologies control when devices are allowed to transmit. This control process is called media access. A major part of thefunction of media access is reducing collisions on the network. Each of the following media access methods has different rules

for media access control:Contention - Carrier Sense Multiple Access/Collision Detection (CSMA/ CD) is an example of a contention system.

Token passing - FDDI and Token Ring networks use token passing.

MAC Addressing

The data link layer is only concerned with physical device addresses, or MAC addresses. MAC addresses are allocated to vendorsby the IEEE, and the vendors assign a unique address to each Network Interface Card (NIC). The format of addresses depends onthe media access method used, which is why they are called MAC addresses. In most LANs, MAC addresses are used to identifythe destination device of a frame.

LLC Data Link Sub-Layer

The LLC data link layer is concerned with transmission synchronisation and connection services.

While the physical layer synchronises the sending and receiving of bits, the data link layer does the same for groups of bits,called frames.


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Transport Layer

The function of the Transport Layer is to accept data from the Session Layer, split it up into smaller

pieces if required, pass these to the Network Layer, and ensure that the pieces all arrive correctly at

the other end.

Under normal conditions, the Transport Layer creates a distinct network connection for each transport

connection required by the Session Layer. However, if the transport connection requires a high

throughput, the Transport Layer might create multiple network connections, dividing the data among

the network connections to improve throughput. On the other hand, if creating or maintaining a

network connection is expensive, the Transport Layer might multiplex several transport connections

onto the same network connection to reduce the cost. In all cases, the Transport Layer is required to

make the multiplexing transparent to the Session Layer.

The Transport Layer is a true source-to-destination or end-to-end layer. In other words, a program on

the source machine carries on a conversation with a similar program on the destination machine, using

the message headers and control messages.

In addition to multiplexing several message streams onto one channel, the Transport Layer musk takecare of establishing and deleting

connections across the network. This requires some kind of naming mechanism, so that process on

one machine has a way of describing with whom it wishes to converse. There must also be a

mechanism to regulate the flow of information, so that a fast host cannot overrun a slow one.


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Session LayerThe purpose of the Session Layer is to facilitate and control communication sessions between service providers and servicerequesters. The Session Layer has functions to establish, maintain, synchronise, and manage communication sessions. Often, ithelps the upper layers identify and connect to the services available on the network.

The two main session layer tasks are:

Dialogue Control

Session AdministrationDialogue Control

There are three types of dialogue that the session layer uses:

Simplex, which allows data to flow in only one direction, Since the dialogue is only one way information can besent, but not responded too, or even acknowledge.

Half duplex, which allows data to flow in two directions, but only one direction at a time.

Full duplex, which lets data flow in both directions simultaneously.

Session AdministrationSession administration covers connection establishment, data transfer, and connection release.

Connection Establishment

Connection establishment involves the following:

Verifying user login names and passwords. Establishing connection identification numbers.

Agreeing which services are required and for how long.

Determining what entity begins the conversation.

Co-ordinating acknowledgement numbering and retransmission procedures.

Data Transfer

Data transfer involves the following:

Actual data transfer.

Acknowledgement of receipt of data, including negative acknowledgement when data is not received.

Resumption of interrupted communication, when required.

Connection Release

Connection release is the task of ending a communication session. Connection release occurs either by agreement of bothparties, or when the connection is broken off for some reason. Entities recognise a lost connection when they do not receive anacknowledgement of data received.


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Presentation Layer

The Presentation Layer transforms data into a format comprehensible to both the sending and the receivingapplication. This format is known as transfer syntax. The Presentation Layer can also compress or expand, andencrypt or decrypt, data.

The main functions of the Presentation Layer are:

TranslationEncryption.

Translation

Computer manufacturers and standards organisations have created rules for converting data (1s and 0s) into aformat that humans can understand. However, these rules often conflict with each other, and translationbetween different rule sets can be required. The following are types of translation:

Bit Order Translation

Bit order translation determines issues such as how many bits constitute a discrete piece of data, and in whatorder bits should be counted.

Byte Order Translation

Byte order translation does the same as bit order translation when different systems use different methods ofgrouping and interpreting bytes.

Character Code Translation

Character code translation translates between different character sets, such as the ANSI standard AmericanStandard Code for Information Interchange (ASCII), and Shift-JIS for Japanese characters. Peer presentation layerprocesses can agree, for example, that the sending and the receiving processes will translate their nativecharacter code into a third, mutually comprehensible, code.

File Syntax Translation

File syntax translation translates file formats between the multitude of local and network Operating Systems (OS)in existence. Network OSs are often required to extract the data and file characteristics from one file system andconvert them for another file system.


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Application LayerThe Application Layer provides the protocols necessary to perform and support network services. These services include the file,print, message, application, and database services. The application layer functions include:

Service advertisement

Service use

Service AdvertisementServers advertise to clients the services they offer. Service advertisement can be either active or passive. With active serviceadvertisement, each server sends out periodic messages to announce availability of its services. Network clients collect theadvertisements and build tables of available services.

Networks which use active service advertisement specify a time interval for the validity of a service. For example, if the timeinterval is five minutes, and a particular service is not advertised in the last five minutes, the client removes that service from itsservice table. Servers perform passive service advertisement by periodically registering their available services with a directory.Clients simply check this directory to find out about the services that are available on each server.

Service Use

Before a network service can be used, it has to be available to the local computersOS. Service use simply means how an OSgains access to a service. The following are service use methods:

OS call interception

Remote operation

Collaborative.

OS Call Interception

With OS call interception, the local OS is completely unaware that the service it is requesting is coming from a network server. Aspecial piece of software intercepts the service request before it reaches the local OS, and sends out a request for the networkservice.

Remote Operation

With remote operation, the local OS is aware of the network and is responsible for submitting service requests. However, theserver is unaware of the client, and treats all requests as if they were of local origin.

Collaborative

With the collaborative method, both service requester and service provider recognise each othersexistence, and collaborate tocoordinate service use.

This method is usually required in peer-to-peer collaborative computing. The collaborative method involves both computerssharing processing capabilities to accomplish a single task.


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Network DevicesAlthough networks started out small, their potential was soon discovered and they started to grow at an alarming rate. Thisgrowth led to the need for better ways of transmitting data across networks. The repeater was the first in a long line of deviceswhich helped to make networks bigger, faster, and more efficient.

Repeater

A repeater takes incoming signals, amplifies them and sends them back into the network without having modified it. This deviceoperates at the physical layer (layer 1of the OSI Reference Model). It connects LAN cables together, thus extending the length ofthe network.

Repeaters are necessary because of the problem of attenuation. Electromagnetic waves become weaker, or attenuate,as theytravel along a transmission medium. A repeater solves this problem by regenerating a degraded signal, thus allowing data to betransmitted along the extra length of network without loss of quality.

The disadvantage of repeaters is that they have no filtering capabilities; they simply pass on any data they receive to the wholenetwork. This can cause congestion on a busy network.

Hub

A hub is a device where many cables converge can also be called a concentrator or a multiport repeater. A hub can beconsidered as a class of repeater because it passes on signals from one media segment to another.

Network Devices (contd)Switch/Bridge

A switch or bridge operates at Layer 2 (Data Link Layer). It connects two network segments similar functions to that of arepeater, except for additional filtering capabilities, thus helping to reduce the network load.

A switch filters and forwards data based on MAC addresses (Media Access Control), which uniquely identify each computer on anetwork, tolearnover time what computer belongs to what segment of the network. This information is stored in the switchs

switching table. The switch uses this information to filter data it receives. If the switch knows a packetsdestination segment, itforwards the packet to that segment only; if itdoesnt,it forwards the packet to all segments.

Router

A router operates at Layer 3 (network layer). A router uses network addresses to identify the destination network of a packet,and only passes on packets with a destination network address.

Routers uses routing tables (metrics) to determine the optimum path for data to travel between two networks. However,routing tables contain more Information, such as the cost of sending a packet along a

particular path. With this Information, a router can choose the best route for a packet to reach its destination, taking intoaccount cost and availability of paths. A disadvantage of routers is that they can be slower than switches.

As a general comment regarding the OSI model, the higher up you go the more intelligent the device but the slower theprocessing. A switch, which is a layer 2 device, is traditionally faster at processing frames than a router is at processing packets.However, a router enables a network to be logically broken up, and is thus more intelligent than a switch.


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The data units flowing from the application layer through the different TCP/IP layers,

is modified, and some headers are added at each layer.

The port number is added at the application layer to identify one of the protocols used,each protocol has a well known port number and other protocols are assigned a

random port numbers.

In the transport layer the transport protocol is chosen, it could be TCP, UDP or others,

and the protocol number is added to identify each of these protocols.

Also, in the internet layer the type code will identify the protocol used by the internet

layer, for example IP

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The internet protocol, is a network layer protocol, or internet layer (in TCP/IP suite), it

provides the transmission of data blocks, called datagrams, from a source to a

destination, between 2 hosts, where the 2 hosts are identified by a 32 bit address each,

called logical address.

The IP sends the data without establishing a virtual connection first, this is called

connectionless transmission the data is transmitted and routed in the network to reach

the final destination.

IP does not provide reliability, so that the data sent is not acknowledged, other upper

layer are responsible for reliability.

The data to be sent could be long, so that the IP provides fragmentation and

reassembly of long data into different IP packets, the maximum size of an IP packet is

1500 bytes.

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The IP uses the addresses provided in the IP header to send data to their destination.

The selection of a path is called IP routing.

The packets arrive to a router first and then this router chooses a path to send the datathrough it, it routes the packet.

In case the router does not know a route to the destination, it drops the packet. This

rarely happens as each router have default routes defined.

The router may also drop the packet in case of errors, after analyzing the checksum.

The routers also decrease the TTL ( time to live field), and if this value is equal to

zero, the packet is also dropped.

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The IP header is divided into many fields which are:

Version, the version, 4 bits, it identifies the version of the protocol, the range is between 0 and 15,value 4 is used for IP4, and 6 is for IP6.

Header length. The header length identifies the number of 4 octets group in the header, it is a pointer tothe beginning of data, it is usually equal to 5, which is for normal header without option, when option is

present, the value of the header length is incremented.

Type of service, this field specifies the priority of the IP datagram, it is newly used.

Total length, 16 bits number that identifies the total length of the IP datagram, which are less than 1500bytes ( the Ethernet frame size), some datagrams could be segmented if they cannot be handled byintermediate routers.

Identification, 16 bits number, assigned by the sender that makes with the sender IP address a uniquenumber used in assembling the fragmented datagrams.

Fragment offset, 13 bits number, used in fragmented datagrams to identify the displacement of thissegment from the beginning of the datagram.

Flags, 3 bits are assigned for flags the low order bit used to identify the last fragment, when set to 0, thehigh order is set to prevent the datagram from being fragmented even it exceeds the size of theintermediate network, it will be discarded but not fragmented.

Time To Live, it is a count, in seconds, set by the sender to specify the time the datagram could stay inthe network before it is discarded.

Protocol, this field is used to identify the higher layer protocol, protocols of the transport layer mainlythe TCP ( value equal to 6) or UDP (value equal to 17), or other protocols used by the transport layer,where each one has a unique number.

Checksum, used to provide assurance that the header has not been corrupted during transmission. Thechecksum is used for the header only (not the data) including the checksum itself.

Addresses, the 32 bits source and destination addresses, that identify the sender and the receiver or the 2communicating hosts.

Options, the presence of the option field is indicated by the value of the length header, the options mayinclude more routing information

Pad, it is added to make the total number of octets divisible by four, as the length of the option couldvaries and it is not necessary equal to 4 bytes, the pad bytes are equal to zeros.

Data, the data field contains the upper layer data unit, which is the transport layer header and data.

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The IP address is a 32 bit number that identifies a host on the network.

Each host has a unique IP address, which is composed of 4 bytes or octets.

The address is normally represented in dotted decimal notation, by

representing the four octets in decimal form separated by a dot.

for example:11000001. 10100000. 00000001. 00000101 represented as

193.160.1.5

The address is divided into two parts:

The network ID, which identifies all hosts located on the same network,

called inter network and has a unique network ID.

The host ID which identifies a unique host within the inter network

For example, 193.160.1.0 is the network ID, and 193.160.1.5 is a host ID

within this network

The network IDs are assigned and controlled, by the Internet Assigned

Numbers Authority (IANA).

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The binary values are either assigned to 0 or 1.

The bits start from bit 0 on the right until bit 7 on the left, so bit n is converted

to decimal by multiplying its value, 0 or 1, by 2^n.

For example if bit 2 is equal to 1, then the decimal value is 1*(2^2) = 4, if it is0 then the result would be 0*(2^2)= 0.

So the binary values are converted to their decimal values, and then the

decimal values of the whole octet is simply the sum of the 8 bits decimal

values.

For example : 1001 = (2^3)*1 + (2^2)*0 + (2^1)*0 + (2^0)*1 = 8+0+0+1 = 9.

The maximum decimal value for an octet is when all the bits are set to 1,

where their sum would be equal to 255.

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The IP address is divided into the network ID, or network prefix, and the host ID.

The IP addresses are distributed into different classes that differ by the number of host they could provide. There aremainly 5 classes.

The class of the IP address is determined by the high order (left-most) bits.

Class A addresses, are assigned to networks with a large number of hosts, the network ID in class A is defined by thefirst byte, the high order bit in class A must be always 0, so the range of network addresses varies from 00000000 = 0to 01111111= 127, so it provides (128-2) possible networks, the 0 is not used and 127 is reserved for diagnosis(loopback test).

Class B addresses are assigned to networks of medium hosts number, the network ID is defined by the first two bytes.In class B the first 2 high order bits must be equal to 10, and so the range of network addresses varies between 128and 191, there are 16 bits for network addresses, the first two bits are assigned 10 so there are 14 bits remaining giving(2^14 = 16384 ) different networks.

Class C addresses are assigned to networks with small number of hosts, the first 3 bytes identify the network address inclass C, the first 3 high order bits are 110, giving a network addresses range from 192 to 223, there are 24 3 first bits= 21 bits resulting in 2^21 or 2097152 networks.

Class D addresses are used for multicast groups.

The multicast group may contain one or more hosts.

The first 4 high order bits are given the value 1110, the remaining bits identify the specific group, in which the hostparticipates.

The addresses are in the range from 224 to 239.

There are no hosts bits in the multicast, packets are passed to a subset of hosts, where hosts, registered in the multicastoperation, receive the packet.

Class E is reserved for future use, the first 5 high order bits are set to 11110.

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The IANA, also created Private IP addresses.

These addresses can be used in private networks, but they are not routable

through the internet.

Each address of these reserved addresses, could be used by one or more

enterprise for its inter network, even a network ID will not remain unique, this

will not cause a problem because these addresses are never injected into the

global internet routing system.

When an organization wishes to get global internet access it needs to use a

Network Address Translator (NAT).

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Some of the global IP addresses are assigned for special use by the Internet

Assigned Number Authority, (IANA).

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For the subnet address scheme to work, every machine on the network must

know which part of the host address will be used as the subnet address.

This is accomplished by assigning a subnet mask to each machine.

A subnet mask is a 32 bit value that allows the recipient to distinguish the

network ID from the host ID.

The 32 bit subnet mask are composed of 1s and 0s, the 1s represent the portion

that refers to the network address, the 0s represent the portion that refers to the

host address.

For example 172.168.10.54 Class B address, so the subnet mask is

255.255.0.0, when the IP is ANDed, with the subnet mask the result would be

172.168.0.0 which is the network ID.

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A company with many internal networks, needs a network address to each of

its internal network.

Routing tables were becoming too large to manage.

To overcome this problem subnetting was initially introduced. It provides a

solution by assigning each organization one network number, and then the

organization is free to assign a distinct subnet networks to each of its internal

networks.

For example, consider the Class B address 160.30.0.0, the default subnet is

255.255.0.0, after subnetting, 160.30.0.0/24 indicates that the first 24 bitsidentify the subnet address so this method has provided 8 additional bits to

divide the network, actually 2^8 1 (broadcast) = 255 different subnet are

available.

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The customized mask allows to divide the network to more subnetworks, each

with a defined number of hosts.

The 160.30.0.0/24, assigns 8 more bits for subnets, so it generates 2^8 = 256subnets.

For example, 160.30.0.0/24 have the following subnets:

160.30.0.0, 160.30.0.1, 160.30.0.2..160.30.0.254, 160.30.0.255.

And within each subnet there is a specific number of hosts.

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VLSM, Variable length subnet mask, is used when there is need to have

different subnets in the network, each with a different number of hosts.

For example if the network ID is 160.40.0.0, the default subnet is 255.255.0.0,after subnetting, the network is divided into many subnets, for example

160.40.144.0 subnet 255.255.252.0, this provides 64 subnets with 1024 -2

hosts in each.

When a less number of users is needed, the VLSM provides the solution

without wasting a hole subnet.

For example between the two router, there is need for two addresses for thetwo interfaces connected, so the number of needed hosts is 2. using

160.40.152.0 with subnet 255.255.255.252 provides 2^22 = 2 hosts.

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The Network address translation NAT, is used to translate the private inside IP addresses to aglobal outside address.

NAT conserves the internet global address space, and it also increases network security byhiding internal IP addresses from external network.

So by using NAT all the hosts inside the network are translated to one public IP address to theoutside network.

The mapping method could be:

Static NAT, allows one to one mapping, each host in the network is mapped to one internet IPaddress.

Dynamic NAT, designed to map an unregistered IP address to one registered IP address, froma pool of registered IP addresses.

Network Address Port Translation (NAPT) is the most popular type of NAT configuration, itis a form of dynamic NAT that maps multiple unregistered IP addresses to a single registeredIP address by using different ports, it is also known as Network Address Port Translation(NAPT)

For example the private addresses 192.168.02. and 192.168.0.3 both send packets from sourceport 1108, A NAPT router may translate these to single public IP address 200.200.160.1 andtwo different ports 31001 and 31002.

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The IP V4 was defined at a time where a few number of computer networks existed,

the 32 bit address was sufficient and allowed over a million networks.

Today the global internet is growing exponentially, and all the network addresses willbe assigned. So there is a need for more addresses.

In addition, the new internet applications, which deliver audio and video need to

deliver data at regular intervals, such information should flow in through the internet

without disruption and changing routes.

The security is also interesting, in IP 6 the packet coming from the host indicated in

the source address, and it cannot be coming from a host other than that indicated inthe source as in IP 4 called spoofing.

So IP v6 provides these solutions, it allows a sufficient number of addresses in the

future, by increasing the size of the IP packet from 32 bits to 128 bits, it will provide

6*(10^23) unique addresses per square meter of the surface of the earth.

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In addition to the increased address size, in IPv6 some Ipv4 header fields have beenmade optional, which reduces the processing cost of packets.

The IP header options in IPv6 are placed in separate optional headers, and most ofthese optional header are not examined and processed by any router on the path. Thissimplifies and speeds up router processing.

IPv6 provides labeling of packets for which the sender requests special handling, suchas real time service for voice or video.

Extension to support authentication, are specified

IPV6 provides address auto configuration, a new version of DHCP has beendeveloped for IPv6, auto configuration does not require a manually configured server,actually a host converts its 48 bits MAC to an EUI 64 bits, and combines it with anetwork prefix that it learns from a neighboring router.

IPv6 provides a new concept, an anycast address, where the packet is delivered to oneof a set of nodes.

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The IPv6 packet begins with header, 40 bytes, the IPv6 header is longer than IPv4 but it contains fewer

fields which reduces router processing and speed up routing.

The header is composed of the following fields:

Version, 4 bits identifying the IP version number.

Priority or Traffic class, it allows to distinguish between different classes of priorities, similar to the

type of service in the IPv4.

Flow Label, 20 bits field, that may be used by a source to label a sequence of packets, which are of the

same class.

Payload length, 16 bits field that indicates the length of the IPv6 payload.

Next Header, 8 bits field identifying the type of header following the IPv6 header.

Hop Limit, 8 bits field used to count the number of routers visited, it is decremented by 1 by each node

that forwards the packet, if the value gets 0 the packet is discarded.

Source and Destination addresses, the source and destination 128 bits IPv6 address

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The Next header identifies the type of the header immediately following the IPv6

header.

Additional optional header can be added in IPv6 which are inserted between the IPv6header and the transport layer header.

The next header then could be the TCP data, which indicates that no optional headers

are inserted.

Routing header could be added and inside the routing header there is a next header to

the following data or headers.

Fragment header could be added to identify fragmented packets.

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Each host in a network will be assigned a unique IPv6 address.

The IPv6 address is composed of a prefix that identify the network and a suffix to

identify a particular host on that network.

Three types of IPv6 addressing , Unicast multicast and anycast.

Unicast, the address corresponds to a single destination, the packet will be routed to

reach that infinity

Multicast, the address corresponds to a set of computers, a copy of the packet is sent

to each of these computers.

Anycast, the address corresponds to a set of computers that have the same address

prefix, the datagram is sent and delivered to one of the computers, the nearest one.

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The main types of unicast addresses in IPv6 are global unicast, site- local unicast, and

link- local unicast.

Link Local, it is used on a single network. Link local are used for purposes such asauto addresses configuration, neighbor discovery or when no routers are present. The

same Link local address can be present on different networks.

Routers should not forward any packet with link local address to other links.

Site Local, used for addressing inside a site without the need for global prefix. Routers

should not forward packets with Site Local addresses outside the site.

Global Unicast, these addresses are used for global communication.

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The IPv6 address is 128 bit long, trying to write it in decimal notation will result in a long number of

characters.

To reduce the number of characters, the designers propose to use the hexadecimal notation.

With hexadecimal notation each byte is represented by 2 hexadecimal characters, and each two bytes

are grouped together and separated from another group by a colon.

For example: 105.220.136.100.255.255.25.0.0.18.128.140.10.255.255 is written in hexadecimal as

follows: 69DC:8864:FFFF:FFFF:0:1280:8C0A:FFFF

To further reduce the size, an additional method known as Zero compression is used, replacing

sequences of zeros with two colons, for example

FF0C:0:0:0:0:0:0:B1 is represented as follows FF0C::B1.

The zero compression may be used only once in an address as it replaces an unknown number of 0, theunspecified address 0:0:0:0:0:0:0:0 which cannot be assigned to any node, can be represented as(::).

The address 0:0:0:0:0:0:0:1 or (::1) is used by a node to send an IP datagram to itself, it is the loopback

address

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All ICMP messages begin with the following three common fields:

Type, 8 bits that identify the message type.

Code field, 8 bits that provide more information about the message type.

Checksum field, 16 bits that is used for error detection.

The identifier and Sequence number, are used to match replies to requests.

Optional Data, contains information to be returned to the sender.

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Ping=Packet Internet Groper uses ICMP echo messages to check the physical and

logical connectivity of machines on an internetwork.

The ping is mainly used in discovering and troubleshooting network problems.

The ping uses ICMP message, and it provides a set of commands with specific

options.

The main function of the options used in ping, are as follows:

W, sets the TTL value.

Ther , -s, -j, and k options exercise various form of source routing.

V option that identify the type o service.



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Hosts on the network must know each others MAC address or hardware address in

order to communicate.

Address resolution is the process of mappinghostsIP address to its hardware address.

The Address resolution protocol is responsible for obtaining MAC addresses for hosts

in a network. It uses a local broadcast of the destination IP address to get the hardware

address of the destination device.

Once the MAC address is obtained the IP and its correspondent MAC address are

stored in ARP cache for a period of time, this is called a dynamic entry.

This ARP cache is checked for an IP address before initiating an ARP request

broadcast.

The IP / MAC addresses mapping could be manually entered to the ARP cache, which

are called static entries



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The source device knows the IP address of the destination, but it wants its hardware

address in order to send the packet.

If the ARP cache does not have an entry for the destination IP, the sender generates anARP request.

The ARP request is a broadcast message, all local devices receive it and check it with

their IP addresses.

The host whose IP address matches the destination address in the ARP request,

generates an ARP reply providing its hardware address to the sender.



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The Transmission Control Protocol (TCP) is one of the core protocols of the

Internet protocol suite, often simply referred to as TCP/IP. Using TCP,

applications on networked hosts can create connections to one another, over

which they can exchange streams of data using Stream Sockets. The protocol

guarantees reliable and in-order delivery of data from sender to receiver. TCP

also distinguishes data for multiple connections by concurrent applications

(e.g., Web server and e-mail server) running on the same host.

In the Internet protocol suite, TCP is the intermediate layer between the

Internet Protocol (IP) below it, and an application above it. Applications often

need reliable pipe-like connections to each other, whereas the Internet

Protocol does not provide such streams, but rather only best effort delivery

(i.e., unreliable packets). TCP does the task of the transport layer in thesimplified OSI model of computer networks. The other main transport-level

Internet protocol is UDP.



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Applications send streams of octets (8-bit bytes) to TCP for delivery through the

network, and TCP divides the byte stream into appropriately sized segments

(usually delineated by the maximum transmission unit (MTU) size of the data link

layer of the network to which the computer is attached). TCP then passes the

resulting packets to the Internet Protocol, for delivery through a network to theTCP module of the entity at the other end.

Connection establishment

To establish a connection, TCP uses a three-way handshake. Before a client attempts

to connect with a server, the server must first bind to a port to open it up for

connections: this is called a passive open. Once the passive open is established, a

client may initiate an active open. To establish a connection, the three-way (or 3-

step) handshake occurs:

The active open is performed by sending a SYN to the server.

In response, the server replies with a SYN-ACK.

Finally the client sends an ACK (usually called SYN-ACK-ACK) back to the server.



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Depending on implementation, the DHCP server has three methods of

allocating IP-addresses:

manual allocation, where the DHCP server performs the allocation based on a

table with MAC address - IP address pairs manually filled by the serveradministrator. Only requesting clients with a MAC address listed in this table

get the IP address according to the table, the manual allocation method

provides a permanent allocation of an IP address.

automatic allocation, where the DHCP server permanently assigns to a

requesting client a free IP-address from a range given by the administrator.

dynamic allocation, the only method which provides dynamic re-use of IP

addresses. A network administrator assigns a range of IP addresses to DHCP,

and each client computer on the LAN has its TCP/IP software configured to

request an IP address from the DHCP server when that client computer'snetwork interface card starts up. The request-and-grant process uses a lease

concept with a controllable time period. This eases the network installation

procedure on the client computer side considerably.



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DHCP requests

Whenever a computer comes on line, it checks to see if it currently has an IP addressleased. If it does not, it requests a lease from a DHCP server. Because the clientcomputer does not know the address of a DHCP server, it uses 0.0.0.0 as its own IP

address and 255.255.255.255 as the destination address. Doing so allows the client tobroadcast a DHCPDISCOVER message across the network. Such a message consistsof the client computer's MAC address.

The client selects a configuration out of the DHCP "Offer" packets it has received andbroadcasts it on the local subnet. Again, this client requests the 160.30.20.150 addressthat the server specified. In case the client has received multiple offers it specifies theserver from which it has accepted the offer.

DHCP acknowledgement

When the DHCP server receives the DHCPREQUEST message from the client, itinitiates the final phase of the configuration process. This acknowledgement phaseinvolves sending a DHCPACK packet to the client. This packet includes the leaseduration and any other configuration information that the client might have requested.At this point, the TCP/IP configuration process is complete.

The server acknowledges the request and sends the acknowledgement to the client.The system as a whole expects the client to configure its network interface with thesupplied options.



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Routers can be configured to interact to act as relay agent, to allow DHCP servers to

serve configuration requests from remote network.

Even the DHCP packets are broadcast packets, routers that conform RFC 1542 canrelay these packets to a remote network.

The router checks the gateway IP address field it is 0.0.0.0, then the router fills the

field with its own IP address.

After receiving the DHCP discover, the server sends DHCP Offer directly to the

gateway address, which will in turn relay the message to the client.



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The DHCP message format is composed of the following fields.

OP, 8 bits operation field to identify the type of the message.

HTYPE, 8 bits identifying the hardware type, for e.g. Ethernet.

HLEN, 8 bits providing the length of the hardware address in the header.

HOPS, 8 bits identifying the number of hops or routers the packet has passed through.

TRANSACTION ID, 32 bits generating an integer to match responses with requests.

SECONDS, 16 bits field, it is defined as the number of seconds elapsed since a client began an attempt

to acquire or renew a lease. This may be used by a busy DHCP server to prioritize replies when

multiple client requests are outstanding.

Client IP ADDRESS, 32 bit IP address, used when the client knows its IP address.

YOUR IP ADDRESS, 32 bits IP address, filled by the server to offer an IP address for the client, if the

client IP address is 0s.

ROUTER IP ADDRESS, 32 bits IP address, set to 0s by the client, when the request passes through a

router, the router records its IP address in this field.

CLIENT HARDWARE ADDRESS, providing the client hardware or MAC address.

SERVER HOST NAME, 64 bytes, optional field providing the server name if it is known, or it is set to

0.

BOOT FILE NAME, 128 bytes, could be set to 0, or providing a bootable filename.

OPTIONS, variable length, used to identify a message type, of the following DHCP messages,

DHCPDISCOVER, DHCPOFFER, DHCPREQUEST, DHCPDECLINE, DHCPACK, DHCPNACK,

DHCPRELEASE.



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In a network there could be a server providing the clock reference, with a peer,

another server with the same stratum level and client which is asking a clock

reference.

Client can asks for a clock reference using Direct pooling, where it can asks one or

more server, and uses the most accurate clock. Or by receiving a broadcasts from NTP

servers passively.



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Hypertext Transfer Protocol (HTTP) is a method used to transfer or convey

information on the World Wide Web.

Its original purpose was to provide a way to publish and retrieve HTML pages, (HyperText Markup Language), the standard language for writing web documents.

Resources to be accessed by HTTP are identified using Uniform Resource Locator (

URLs) using the http: or https URL schemes.

e.g. http://www.ngentelecom.com



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The HTTP message format is composed of the following parts:

Command, could be GET to retrieve data, POST to place data on the server.

The URL, containing:

Protocol, identifying the used protocol, could be HTTP or FTP or others.

HTTP server domain name.

Path name.

File name.



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When client clicks on a link, e.g. www.ngen.com, the following steps occur:

The browser determines the URL.

The browser asks DNS for the IP address for www.ngen.com

DNS replies with the IP address

TCP connection is established with the server

The client then sends a GET http:// www.ngen.com

The server sends the main or default page

The TCP connection is then released

The browser displays all the text in the main page

Then the browser fetches and displays all images in the default page, for each image,icon, or photos, a new TCP connection is established



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The SMTP, simple mail transfer protocol, is used to facilitate the exchange of

electronic message between users on a network.

SMTP is an application layer protocol, that uses the port number 25.



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The message is created and sent by a user.

The agent is the portion of the client that interfaces with the user , it accepts elements

of the message in two parts, the header part and message part.

The header contains the required fieldsTo, Reply To, CC.

The message part is the text.

The client is responsible to establish a TCP connection with each remote SMTP

server, and send the messages.

The SMTP server places each received message in the corresponding queue of the

appropriate mailbox.



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The SMTP uses the following commands.

HELLO, after sending the welcoming message the client sends the HELLO command

to the server indicating theclientsidentity.

MAIL, mail command is the first command in the process after connection

establishment, the mail command is used to identify the argument.

RCPT, recipient command is used to identify an individual recipient of the mail.

DATA, data command informs the SMTP server that the data will be sent now.

SEND, the same as RCPT command except that the message will be sent to a terminal

instead of a mailbox.

QUIT, is used when the client finishes sending data to inform the server that the

connection will be closed.



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TELNET (TELetypeNETwork) is a network protocol used on the Internet or local

area network (LAN) connections. It was developed in 1969.

TELNET is a client-server protocol, based on a reliable connection-oriented transport.

Typically this is TCP port 23.

TELNET, does not encrypt any data sent over the connection (including passwords),

anybody who has access to a router, switch, or gateway located on the network

between the two hosts where TELNET is being used can intercept the packets passing

by and easily obtain login and password information.



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telnet provides a standardized interface through which, a program on one host ( telnet

client) accesses the resources of another host (telnet server).

As if the client were a local terminal connected to the server.

telnet is also used for logging into bridges, routers and other network devices for

management and configuration.



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File transfer protocol is the protocol that allows the transfer of files between two

hosts.

FTP is also used by programs, or user applications, that allow humans to easily

interact with remote servers.

FTP requires authentication, so the clients have to send their login ID and passwords

to the server before file transfer.



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VLANs allows to group different computer from different location into

one network, as they are physically connected.

VLANs allow logical network topologies to overlay the physical switched

infrastructure such that any arbitrary collection of LAN ports can be

combined into an autonomous user group or same broadcast domain, it

enables switches to create multiple broadcast domain.

VLANs also improve security by isolating groups. High-security userscan be grouped into a VLAN, possible on the same physical segment, and

no users outside that VLAN can communicate with them.

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Each VLAN operates in the same manner as a switch without VLAN, it

provides address learning filtering and loop avoidance.

Traffic originating from one VLAN will be flooded only to the ports

belonging to that VLAN.

One VLAN can be distributed into many switches, this need a trunk line

to be used, by configuring one port as a trunk, this trunk can carry traffic

for many VLANs, separated by a tags.



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Static VLAN are the usual way of creating VLANs, they are the

most secure.

The switch port when assigned to a VLAN will always be a

member of that VLAN until it is manually reassigned to another

VLAN.

Dynamic VLAN, provides the node with automatic assignment

using a management software and a database.

The dynamic assignment could be based on MAC addresses or

Protocols, by providing a database with the MAC addresses or

protocols and their VLAN assignments.

The switch will look up the database and dynamically assigns the

node to the correspondent VLAN.



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Trunk links: Trunk s can carry multiple VLANs and originally gained

their name after the telephone system trunks that carry multiple telephone

conversations.

A trunk link is a point to point link between two switches, between a

switch and router, or between a switch and server. These carry the traffic

of multiple VLANs.

A user-defined ID is assigned to each frame to identify its VLAN

membership. Sometimes people refer it as a VLAN IDorcolor.



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There are three major methods of frame tagging,

Cisco proprietary Inter-Switch Link (ISL) This is propriety to Ciscoswitches, anditsused to Fast Ethernet and Gigabit Ethernet links only

IEEE 802.1Q, Created by the IEEE as a standard method of frame

tagging, it actually a field into the frame to identify the VLAN. The

connection between switches of two different brands the 802.1 q should

be used for the trunk to work.

3Com VLT (Virtual LAN Trunk). This is propriety to 3Com switches.



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Redundant links between switches are a good idea because they

prevent complete network failure, if one link stops working.

Even though redundant links are helpful, they may cause a lot of

problems, because frames could be flooded through all the

redundant links creating loops.

If more than one open path were to be active at once then there

would be several problems.First, a broadcast storm caused by broadcast packets looping

between switches would reduce bandwidth

Second, the traditional source-based location system (filtering

database) used by switches would fail to operate correctly



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The main task of the Spanning Tree Protocol is to prevent network loops

from occurring on layer 2 networks (switches and bridges).

Spanning-Tree Protocol is a link management protocol that provides path

redundancy while preventing undesirable loops in the network. For an

Ethernet network to function properly, only one active path can exist

between two stations.

STP has been standardized by IEEE 802.1D.

As the name suggests it finds a spanning tree within the mesh network

formed, and disables the links not part of that tree.



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All switches that participate in a Spanning Tree protocol gather

information on other switches by sending and receiving messages.

Each switch has a unique identifier (ID) and a configurable priority

number; both of these numbers make up the Bridge Identification or BID.

The BID is used to elect a root bridge based upon the lowest priority

number; if this is a tie then the numerically lowest ID wins

Spanning Tree messages are called BPDUs (bridge protocol data units).

The result of message exchange should:

Elect a root switch for a stable tree network topology

Elect a designated port for every LAN segment

Remove loops in network by placing redundant ports in a backup

state



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Per-VLAN Spanning Tree (PVST)

In Ethernet switched environments where multiple Virtual LANs

exist, spanning tree can be deployed per Virtual LAN. Cisco's name

for this is per VLAN spanning tree. Cisco's name for this is per

VLAN spanning tree (PVST and PVST+ which is the default

protocol used by Cisco switches). Both PVST and PVST+

protocols are Cisco proprietary protocols.

Rapid Spanning Tree Protocol (RSTP)

RSTP is an evolution of the Spanning Tree Protocol it was

introduced in the extension IEEE 802.1w, and provides for faster

spanning tree convergence after a topology change.

The Multiple Spanning Tree Protocol (MSTP), originally defined in

IEEE 802.1s and later merged into IEEE 802.1Q-2003, defines an

extension to the RSTP protocol to further develop the usefulness of

virtual LANs (VLANs). This "Per-VLAN" Multiple Spanning Tree

Protocol configures a separate Spanning Tree for each VLAN group

and blocks the links that are redundant within each Spanning Tree.



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Routing tables instruct the routers how to route packets.

For each address there is an entry in the routing table, which are calledroutes.

Routers may also have a default route to external destinations that are not

present in the routing table.

There are two types of routing, which are:

Dynamic routing.

Static routing



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In static routing routes are added manually in the routersrouting table.

Static routing decreases the routers CPU usage and provides security

because the administrator can choose to allow routing access to certain

networks only.

But this type of routing would be difficult to implement in in large

networks where maintaining it needs a lot of time.



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The distance vector algorithm operates in a very simple manner. Whena node first starts, it only knows of its immediate neighbors, and thedirect cost involved in reaching them. The list of destinations, the totalcost to each, and the next hop to send data to get there, makes up the

routing table, or distance table.Each node, on a regular basis, sends to each neighbor its own currentidea of the total cost to get to all the destinations it knows of. Theneighboring nodes examine this information, and compare it to whatthey already 'know'; anything which represents an improvement onwhat they already have, they insert in their own routing tables. Overtime, all the nodes in the network will discover the best next hop for alldestinations, and the best total cost.

The Link State algorithm. Each router independently determines thebest route from itself to every other node using Dijkstra's algorithm, bybuilding a tree with the current node itself as the root, and containingevery other node in the network. It starts with a tree containing onlyitself. Then, one at a time, from the set of nodes which it has not yetadded to the tree, it adds the node which has the lowest cost to reach anadjacent node which already appears in the tree. This continues untilevery node appears in the tree.



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RIP uses a single routing metric (hop count) to measure the distancebetween the source and a destination network. Each hop in a path fromsource to destination is assigned a hop count value.

When the router receives an update message that containing a route to adestination with a better hop count, the routing table is updated.

Router sends update message to neighbors every 30 sec

A router expects to receive an update message from each of its neighborswithin 180 seconds in the worst case

If router does not receive update message from neighbor X within thislimit, it assumes the link to X has failed and sets the correspondingminimum cost to 16 (infinity)

RIP messages are sent using UDP transmission protocol with port number520, encapsulated over IP packets.



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Despite RIP's age and the emergence of more sophisticated routing

protocols, it is far from obsolete.

RIP is mature, stable, widely supported, and easy to configure.

Its simplicity is well suited for use in stub networks and in small

autonomous systems that do not have enough redundant paths to warrant

the overheads of a more sophisticated protocol.



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Early routing protocols such as RIP v1 were all distance vector protocols.

There are many distance vector routing protocols in use today such as RIPv2, IGRP, and the hybrid routing protocol EIGRP.

As networks have grown larger and more complex, the limitations of

distance vector routing protocols have become apparent.

Routers that use a distance vector routing protocol learn about the

network topology from the routing table updates of neighbor routers.

Bandwidth usage is high because of the periodic exchange of routing

updates, and network convergence is slow which results in poor routing

decisions.



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Module IV: 3G & 4G Transmission/


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ATM:

# of cell required = 500M/48

= 524288000/48

= 10922667

#byte of header = 10922667 * 5

= 54613333

= 53333 Kbyte

IP:

# of packet required = 500M/1480

= 524288000/1480

= 354249

#byte of header = 354249 * 20

= 7084973

= 6919 Kbyte


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This is a great opportunity for us on the access side but it creates also a greatopportunity on the infrastructure side both from an IP and Optical perspective

The opportunity for us is to build a best of class IP network that can keep up withthese types of growth and changes and at the same time provides a best of classconverged solution which will save our customers on both CAPEX and OPEX

IP technology is deployed is over 500 customers around the world in both Mobileand Fixed networks.

There are 3 major technological differentiations:

1. IP technology provides our customers with an elegant migration path from legacyt k t IP It l id th ith t fl ibilit th d l

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