Q: 1 the OSI Reference Model: 1. The Physical Layer:

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1 Q: 1 the OSI Reference Model: This model is based on a proposal developed by the International Standards Organization (ISO) as a first step toward international standardization of the protocols used in the various layers. The model is called the ISO OSI (Open Systems Interconnection) Reference Model because it deals with connecting open systems-that is, systems that are open for communication with other systems. We will just call it the OSI model for short. The OSI model has seven layers. The principles that were applied to arrive at the seven layers can be briefly summarized as follows: 1. A layer should be created where a different abstraction is needed. 2. Each layer should perform a well-defined function. 3. The function of each layer should be chosen with an eye toward defining internationally standardized protocols. 4. The layer boundaries should be chosen to minimize the information flow across the interfaces. 5. The number of layers should be large enough that distinct functions need not be thrown together in the same layer out of necessity and small enough that the architecture does not become unwieldy. Below we will discuss each layer of the model in turn, starting at the bottom layer. Note that the OSI model itself is not network architecture because it does not specify the exact services and protocols to be used in each layer. It just tells what each layer should do. However, ISO has also produced standards for all the layers, although these are not part of the reference model itself. Each one has been published as a separate international standard. 1. The Physical Layer: The physical layer is concerned with transmitting raw bits over a communication channel. The design issues have to do with making sure that when one side sends a 1 bit, it is received by the other side as a 1 bit, not as a 0 bit. Typical questions here are how many volts should be used to represent a 1 and how many for a 0, how many nanoseconds a bit lasts, whether transmission may proceed simultaneously in both directions, how the initial connection is established and how it is torn down when both sides are finished, and how many pins the network

2 connector has and what each pin is used for. The design issues here largely deal with mechanical, electrical, and timing interfaces, and the physical transmission medium, which lies below the physical layer. 2. The Data Link Layer: The main task of the data link layer is to transform a raw transmission facility into a line that appears free of undetected transmission errors to the network layer. It accomplishes this task by having the sender break up the input data into data frames (typically a few hundred or a few thousand bytes) and transmits the frames sequentially. If the service is reliable, the receiver confirms correct receipt of each frame by sending back an acknowledgement frame. Another issue that arises in the data link layer (and most of the higher layers as well) is how to keep a fast transmitter from drowning a slow receiver in data. Some traffic regulation mechanism is often needed to let the transmitter know how much buffer space the receiver has at the moment. Frequently, this flow regulation and the error handling are integrated. Broadcast networks have an additional issue in the data link layer: how to control access to the shared channel. A special sub layer of the data link layer, the medium access control sublayer, deals with this problem. 3. The Network Layer: The network layer controls the operation of the subnet. A key design issue is determining how packets are routed from source to destination. Routes can be based on static tables that are ''wired into'' the network and rarely changed. They can also be determined at the start of each conversation, for example, a terminal session (e.g., a login to a remote machine). Finally, they can be highly dynamic, being determined anew for each packet, to reflect the current network load. If too many packets are present in the subnet at the same time, they will get in one another's way, forming bottlenecks. The control of such congestion also belongs to the network layer. More generally, the quality of service provided (delay, transit time, jitter, etc.) is also a network layer issue. When a packet has to travel from one network to another to get to its destination, many problems can arise. The addressing used by the second network may be different from the first one. The second one may not accept the packet at all because it is too large. The protocols may differ, and so on. It is up to the network layer to overcome all these problems to allow heterogeneous networks to be interconnected. In broadcast networks, the routing problem is simple, so the network layer is often thin or even nonexistent. 4. The Transport Layer: The basic function of the transport layer is to accept data from above, split it up into smaller units if need be, pass these to the network layer, and ensure that the pieces all arrive correctly at the other end. Furthermore, all this must be done efficiently and in a way that isolates the upper layers from the inevitable changes in the hardware technology. The transport layer also determines what type of service to provide to the session layer, and, ultimately, to the users of the network. The most popular type of transport connection is an error-free point-to-point channel that delivers messages or bytes in the order in which they were sent. However, other possible kinds of transport service are the transporting of isolated messages, with no guarantee about the order of delivery, and the broadcasting of messages to multiple destinations. The type of service is determined when the connection is established. (As an aside, an error-free channel is impossible to achieve; what people really mean by this term is that the error rate is low enough to ignore in practice.) The transport layer is a true end-to-end layer, all the way from the source to the destination. 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 the lower layers, the protocols are between each machine and its immediate neighbors, and not between the ultimate source and destination machines, which may be separated by many routers. The difference between layers 1 through 3, which are chained, and layers 4 through 7, which are end-to-end, is illustrated in Fig The Session Layer: The session layer allows users on different machines to establish sessions between them. Sessions offer various services, including dialog control (keeping track of whose turn it is to transmit), token management (preventing two parties from attempting the same critical operation at the same time), and synchronization (checkpointing long transmissions to allow them to continue from where they were after a crash).

3 6. The Presentation Layer: Unlike lower layers, which are mostly concerned with moving bits around, the presentation layer is concerned with the syntax and semantics of the information transmitted. In order to make it possible for computers with different data representations to communicate, the data structures to be exchanged can be defined in an abstract way, along with a standard encoding to be used ''on the wire.'' The presentation layer manages these abstract data structures and allows higher-level data structures (e.g., banking records), to be defined and exchanged. 7. The Application Layer: The application layer contains a variety of protocols that are commonly needed by users. One widely-used application protocol is HTTP (Hyper Text Transfer Protocol), which is the basis for the World Wide Web. When a browser wants a Web page, it sends the name of the page it wants to the server using HTTP. The server then sends the page back. Other application protocols are used for file transfer, electronic mail, and network news. Q2: Transmission Impairments: When a signal is transmitted over a communication channel, it is subjected to different types of impairments because of imperfect characteristics of the channel. As a consequence, the received and the transmitted signals are not the same. Outcome of the impairments are manifested in two different ways in analog and digital signals. These impairments introduce random modifications in analog signals leading to distortion. On the other hand, in case of digital signals, the impairments lead to error in the bit values. The impairment can be broadly categorized into the following three types: Attenuation and attenuation distortion Delay distortion Noise ATTENUATION: Irrespective of whether a medium is guided or unguided, the strength of a signal falls off with distance. This is known as attenuation. In case of guided media, the attenuation is logarithmic, whereas in case of unguided media it is a more complex function of the distance and the material that constitutes the medium. An important concept in the field of data communications is the use of on unit known as decibel (db). To define it let us consider the circuit elements shown in Figure. The elements can be a transmission line, an amplifier, an attenuator, a filter, etc. In the figure, a transmission line (between points P 1 and P 2 ) is followed by an amplifier (between P 2 and P 3 ). The input signal delivers a power P 1 at the input of a communication element and the output power is P2. Then the power gain G for this element in decibels is given by G = 10log 2 P 2 / P 1. Here P 2 / P 1 are referred to as absolute power gain. When P 2 > P 1, the gain is positive, whereas if P 2 < P 1, then the power gain is negative and there is a power loss in the circuit element. DELAY DISTORTION: The velocity of propagation of different frequency components of a signal are different in guided media. This leads to delay distortion in the signal. For a band limited signal, the velocity of propagation has been found to be maximum near the center frequency and lower on both sides of the edges of the frequency band. In case of analog signals, the received signal is distorted because of variable delay of different components. In case of digital signals, the problem is much more severe. Some frequency components of one bit position spill over to other bit positions, because of delay distortion. This leads to intersymbol interference, which restricts the maximum bit rate of transmission through a particular transmission medium. The delay distortion can also be neutralized, like attenuation distortion, by using suitable equalizers.

4 NOISE: As signal is transmitted through a channel, undesired signal in the form of noise gets mixed up with the signal, along with the distortion introduced by the transmission media. Noise can be categorized into the following four types: Thermal Noise Intermodulation Noise Cross talk Impulse Noise The thermal noise is due to thermal agitation of electrons in a conductor. It is distributed across the entire spectrum and that is why it is also known as white noise (as the frequency encompass over a broad range of frequencies). When more than one signal share a single transmission medium, intermodulation noise is generated. For example, two signals f1 and f2 will generate signals of frequencies (f 1 + f 2 ) and (f 1 - f 2 ), which may interfere with the signals of the same frequencies sent by the transmitter. Intermodulation noise is introduced due to nonlinearity present in any part of the communication system. Cross talk is a result of bunching several conductors together in a single cable. Signal carrying wires generate electromagnetic radiation, which is induced on other conductors because of close proximity of the conductors. While using telephone, it is a common experience to hear conversation of other people in the background. This is known as cross talk. Impulse noise is irregular pulses or noise spikes of short duration generated by phenomena like lightning, spark due to loose contact in electric circuits, etc. Impulse noise is a primary source of bit-errors in digital data communication. This kind of noise introduces burst errors. Q4: IP ADDRESSING: An IP address is an address used in order to uniquely identify a device on an IP network. The address is made up of 32 binary bits, which can be divisible into a network portion and host portion with the help of a subnet mask. The 32 binary bits are broken into four octets (1 octet = 8 bits). Each octet is converted to decimal and separated by a period (dot). For this reason, an IP address is said to be expressed in dotted decimal format (for example, ). The value in each octet ranges from 0 to 255 decimal, or binary. Octets are broken down to provide an addressing scheme that can accommodate large and small networks. There are five different classes of networks, A to E. Of the five classes, D and E are dedicated to special purposes, Classes A, B and C are the ones actually assigned for normal (unicast) addressing purposes on IP internetworks Given an IP address, its class can be determined from the three high-order bits. Figure 1 shows the significance in the three high order bits and the range of addresses that fall into each class. For informational purposes, Class D and Class E addresses are also shown.

5 Figure 1 In a Class A address, the first octet is the network portion, so the Class A example in Figure 1 has a major network address of Octets 2, 3, and 4 (the next 24 bits) are for the network manager to divide into subnets and hosts as he/she sees fit. Class A addresses are used for networks that have more than 65,536 hosts (actually, up to hosts!). In a Class B address, the first two octets are the network portion, so the Class B example in Figure 1 has a major network address of Octets 3 and 4 (16 bits) are for local subnets and hosts. Class B addresses are used for networks that have between 256 and hosts. In a Class C address, the first three octets are the network portion. The Class C example in Figure 1 has a major network address of Octet 4 (8 bits) is for local subnets and hosts - perfect for networks with less than 254 hosts. Class 1st Octet Decimal Range 1st Octet High Order Bits Network/Host ID (N=Network, H=Host) A 1 126* 0 N.H.H.H B N.N.H.H C N.N.N.H Usable No. of network ID bits 8-1 = = = 21 Default Subnet Mask Number of Networks (2 7 2) ,382 (2 14 2) 2,097,150 (2 21 2) D Reserved for Multicasting E Experimental; used for research Hosts per Network (Usable Addresses) 16,777,214 (2 24 2) 65,534 (2 16 2) 254 (2 8 2) Note: Class A addresses to cannot be used and is reserved for loopback and diagnostic functions.

6 Understanding Subnetting: Subnetting allows you to create multiple logical networks that exist within a single Class A, B, or C network. If you do not subnet, you are only able to use one network from your Class A, B, or C network, which is unrealistic. Each data link on a network must have a unique network ID, with every node on that link being a member of the same network. If you break a major network (Class A, B, or C) into smaller subnetworks, it allows you to create a network of interconnecting subnetworks. Each data link on this network would then have a unique network/subnetwork ID. Any device, or gateway, connecting n networks/subnetworks has n distinct IP addresses, one for each network / subnetwork that it interconnects. In order to subnet a network, extend the natural mask using some of the bits from the host ID portion of the address to create a subnetwork ID. For example, given a Class C network of which has a natural mask of , you can create subnets in this manner: sub ---- By extending the mask to be , you have taken three bits (indicated by "sub") from the original host portion of the address and used them to make subnets. With these three bits, it is possible to create eight subnets. With the remaining five host ID bits, each subnet can have up to 32 host addresses, 30 of which can actually be assigned to a device since host ids of all zeros or all ones are not allowed (it is very important to remember this). So, with this in mind, these subnets have been created host address range 1 to host address range 33 to host address range 65 to host address range 97 to host address range 129 to host address range 161 to host address range 193 to host address range 225 to 254 HAMMING CODE: Hamming code is a set of error-correction code s that can be used to detect and correct bit errors that can occur when computer data is moved or stored. Hamming code is named for R. W. Hamming of Bell Labs. Like other error-correction code, Hamming code makes use of the concept of parity and parity bit s, which are bits that are added to data so that the validity of the data can be checked when it is read or after it has been received in a data transmission. Using more than one parity bit, an error-correction code can not only identify a single bit error in the data unit, but also its location in the data unit. Calculating the Hamming Code The key to the Hamming Code is the use of extra parity bits to allow the identification of a single error. Create the code word as follows: 1. Mark all bit positions that are powers of two as parity bits. (positions 1, 2, 4, 8, 16, 32, 64, etc.) 2. All other bit positions are for the data to be encoded. (positions 3, 5, 6, 7, 9, 10, 11, 12, 13, 14, 15, 17, etc.) 3. Each parity bit calculates the parity for some of the bits in the code word. The position of the parity bit determines the sequence of bits that it alternately checks and skips. Position 1: check 1 bit, skip 1 bit, check 1 bit, skip 1 bit, etc. (1,3,5,7,9,11,13,15,...) Position 2: check 2 bits, skip 2 bits, check 2 bits, skip 2 bits, etc. (2,3,6,7,10,11,14,15,...) Position 4: check 4 bits, skip 4 bits, check 4 bits, skip 4 bits, etc. (4,5,6,7,12,13,14,15,20,21,22,23,...)

7 Position 8: check 8 bits, skip 8 bits, check 8 bits, skip 8 bits, etc. (8-15,24-31,40-47,...) Position 16: check 16 bits, skip 16 bits, check 16 bits, skip 16 bits, etc. (16-31,48-63,80-95,...) Position 32: check 32 bits, skip 32 bits, check 32 bits, skip 32 bits, etc. (32-63,96-127, ,...) etc. 4. Set a parity bit to 1 if the total number of ones in the positions it checks is odd. Set a parity bit to 0 if the total number of ones in the positions it checks is even. Here is an example: A byte of data: Create the data word, leaving spaces for the parity bits: 1 _ _ Calculate the parity for each parity bit (a? represents the bit position being set): Position 1 checks bits 1,3,5,7,9,11:? _ 1 _ _ Even parity so set position 1 to a 0: 0 _ 1 _ _ Position 2 checks bits 2,3,6,7,10,11: 0? 1 _ _ Odd parity so set position 2 to a 1: _ _ Position 4 checks bits 4,5,6,7,12: 0 1 1? _ Odd parity so set position 4 to a 1: _ Position 8 checks bits 8,9,10,11,12: ? Even parity so set position 8 to a 0: Code word: Finding and fixing a bad bit The above example created a code word of Suppose the word that was received was instead. Then the receiver could calculate which bit was wrong and correct it. The method is to verify each check bit. Write down all the incorrect parity bits. Doing so, you will discover that parity bits 2 and 8 are incorrect. It is not an accident that = 10, and that bit position 10 is the location of the bad bit. In general, check each parity bit, and add the positions that are wrong, this will give you the location of the bad bit. X.25 Network: X.25 is a standard suite of protocols used for packet switching across computer networks. The X.25 protocols works at the physical, data link, and network layers (Layers 1 to 3) of the OSI model. Each X.25 packets contains up to 128 bytes of data. The X.25 network handles packet assembly at the source device, delivery, and then dis-assembly at the destination. X.25 packet delivery technology includes not only switching and network-layer routing, but also error checking and re-transmission logic should delivery failures occur. X.25 supports multiple simultaneous conversations by multiplexing packets and using virtual communication channels. X.25 was originally designed more than 25 years ago to carry voice over analog telephone lines (dialup networks). Typical applications of X.25 today include automatic teller machine networks and credit card verification networks. X.25 also supports a variety of mainframe terminal/server applications. With the widespread acceptance of Internet Protocol (IP) as a standard for corporate networks, many X.25 applications are now being migrated to cheaper solutions using IP as the network layer protocol and replacing the lower layers of X.25 with Ethernet or ATM hardware. ATM (Asynchronous Transfer Mode): ATM is a high-speed networking standard designed to support both voice and data communications. ATM is normally utilized by Internet service providers on their private long-distance networks. ATM operates at the data link layer (Layer 2 in the OSI model) over either fiber or twisted-pair cable. ATM differs from more common data link technologies like Ethernet in several ways. For example, ATM utilizes no routing. Hardware devices known as ATM switches establish point-to-point connections between endpoints and data flows directly from source to destination. Additionally, instead of using variable-length

8 packets as Ethernet does, ATM utilizes fixed-sized cells. ATM cells are 53 bytes in length, that includes 48 bytes of data and five (5) bytes of header information. The performance of ATM is often expressed in the form of OC (Optical Carrier) levels, written as "OC-xxx." Performance levels as high as 10 Gbps (OC-192) are technically feasible with ATM. More common performance levels for ATM are 155 Mbps (OC-3) and 622 Mbps (OC-12). ATM technology is designed to improve utilization and quality of service (QoS) on high-traffic networks. Without routing and with fixed-size cells, networks can much more easily manage bandwidth under ATM than under Ethernet, for example. The high cost of ATM relative to Ethernet is one factor that has limited its adoption to "backbone" and other high-performance, specialized networks. Open Shortest Path Routing: (OSPF) Let us begin our study of feasible routing algorithms with a technique that is widely used in many forms because it is simple and easy to understand. The idea is to build a graph of the subnet, with each node of the graph representing a router and each arc of the graph representing a communication line (often called a link). To choose a route between a given pair of routers, the algorithm just finds the shortest path between them on the graph. The concept of a shortest path deserves some explanation. One way of measuring path length is the number of hops. Using this metric, the paths ABC and ABE in Fig. 5-7 are equally long. Another metric is the geographic distance in kilometers, in which case ABC is clearly much longer than ABE (assuming the figure is drawn to scale). Figure 5-7. The first five steps used in computing the shortest path from A to D. The arrows indicate the working node. However, many other metrics besides hops and physical distance are also possible. For example, each arc could be labeled with the mean queuing and transmission delay for some standard test packet as determined by hourly test runs. With this graph labeling, the shortest path is the fastest path rather than the path with the fewest arcs or kilometers. In the general case, the labels on the arcs could be computed as a function of the distance, bandwidth, average traffic, communication cost, mean queue length, measured delay, and other factors. By changing the weighting function, the algorithm would then compute the ''shortest'' path measured according to any one of a number of criteria or to a combination of criteria.

9 Several algorithms for computing the shortest path between two nodes of a graph are known. This one is due to Dijkstra (1959). Each node is labeled (in parentheses) with its distance from the source node along the best known path. Initially, no paths are known, so all nodes are labeled with infinity. As the algorithm proceeds and paths are found, the labels may change, reflecting better paths. A label may be either tentative or permanent. Initially, all labels are tentative. When it is discovered that a label represents the shortest possible path from the source to that node, it is made permanent and never changed thereafter. To illustrate how the labeling algorithm works, look at the weighted, undirected graph of Fig. 5-7(a), where the weights represent, for example, distance. We want to find the shortest path from A to D. We start out by marking node A as permanent, indicated by a filled-in circle. Then we examine, in turn, each of the nodes adjacent to A (the working node), relabeling each one with the distance to A. Whenever a node is relabeled, we also label it with the node from which the probe was made so that we can reconstruct the final path later. Having examined each of the nodes adjacent to A, we examine all the tentatively labeled nodes in the whole graph and make the one with the smallest label permanent, as shown in Fig. 5-7(b). This one becomes the new working node. We now start at B and examine all nodes adjacent to it. If the sum of the label on B and the distance from B to the node being considered is less than the label on that node, we have a shorter path, so the node is relabeled. After all the nodes adjacent to the working node have been inspected and the tentative labels changed if possible, the entire graph is searched for the tentatively-labeled node with the smallest value. This node is made permanent and becomes the working node for the next round. Figure 5-7 shows the first five steps of the algorithm. To see why the algorithm works, look at Fig. 5-7(c). At that point we have just made E permanent. Suppose that there were a shorter path than ABE, say AXYZE. There are two possibilities: either node Z has already been made permanent, or it has not been. If it has, then E has already been probed (on the round following the one when Z was made permanent), so the AXYZE path has not escaped our attention and thus cannot be a shorter path. Now consider the case where Z is still tentatively labeled. Either the label at Z is greater than or equal to that at E, in which case AXYZE cannot be a shorter path than ABE, or it is less than that of E, in which case Z and not E will become permanent first, allowing E to be probed from Z.