Assignment

Ignou bca cs-68 solved assignment



Course Code : CS-68

Course Title : Computer Networks

Assignment Number :BCA(5)CS-68/Assignment/2011

Maximum Marks : 100 (Weightage = 25%)

Last Date of Submission : 30th April, 2011/30th October, 2011







Question 1: Differentiate the followings with appropriate examples:

(i) Local and Remote Bridges

(ii) Constant Bit Rate and Variable Bit Rate

(iii) Time Division Multiplexing and Frequency Division Multiplexing

(iv) Datagram and Virtual Circuit

(v) Analog and Digital Signal



Ans : A computer network, often simply referred to as a network, is a collection of computers and devices interconnected by communications channels that facilitate communications among users and allows users to share resources. Networks may be classified according to a wide variety of characteristics. A computer network allows sharing of resources and information among interconnected devices.



(i) local and remote bridges

ans : Local bridges: Directly connect local area networks (LANs).



Local bridges are ties between two nodes in a social graph that are the shortest (and often the only plausible) route by which information might travel from those connected to one to those connected to the other. [2] Local bridges differ from regular bridges in that the end points of the local bridge cannot have a tie directly between them and should not share any common neighbors.



The local bridge connection function (herein referred to as local bridge) can connect a Virtual HUB operating on the VPN Server or VPN Bridge and the physical network adapter connected to that server computer on a layer 2 connection, thereby joining two segments which originally operated as separate Ethernet segments into one.

Local bridging enables a computer connected to a Virtual HUB and a computer connected to a physical LAN to communicate freely on an Ethernet level connected, in theory, to the same Ethernet segment, regardless of whether each of them is physically linked to a separate network.

Using a local bridge makes it possible to easily construct a remote-access VPN and site-to-site VPN.The local bridge is a function often used by the PacketiX VPN to make VPN connections. Local bridging is used to connect a virtual network and a physical network on the Ethernet level. This section will explain local bridge concepts, methods for setting them and precautions.



Remote bridges: Can be used to create a wide area network (WAN) link between LANs. Remote bridges, where the connecting link is slower than the end networks, largely have been replaced with routers.A common use of bridges is to connect two(or more) distant LANs. For example, a company might have plants in several cities, each with its own LAN. Ideally, all the LANs should be interconnected, so the complete system acts like one large LAN. This goal can be achieved by putting abridge on each LAN and connecting the bridges pair wise with point to point lines. Various protocols can be used on the point to point lines. One possibility is to choose some standard point-to-point data link protocol such as PPP, putting complete MAC frames in the payload field.



This strategy works best if all the LANs are identical, and the only problem is getting frames to the correct LAN. Another option is to strip off the MAC header and trailer at the Source Bridge and header and tailor can than be generated at the designation bridge. A disadvantage of this approaches that the checksum that arrives at the destination host is not the one computed by the source host, so errors caused by bad bits in a bridge's memory may not be detected



(ii) Constant Bit Rate and Variable Bit Rate

ans : Constant bitrate (CBR) is a term used in telecommunications, relating to the quality of service. Compare with variable bitrate.



When referring to codecs, constant bit rate encoding means that the rate at which a codec's output data should be consumed is constant. CBR is useful for streaming multimedia content on limited capacity channels since it is the maximum bit rate that matters, not the average, so CBR would be used to take advantage of all of the capacity. CBR would not be the optimal choice for storage as it would not allocate enough data for complex sections (resulting in degraded quality) while wasting data on simple sections.



The problem of not allocating enough data for complex sections could be solved by choosing a high bitrate (e.g., 256 kbit/s or 320 kbit/s) to ensure that there will be enough bits for the entire encoding process, though the size of the file at the end would be proportionally larger.



Most coding schemes such as Huffman coding or run-length encoding produce variable-length codes, making perfect CBR difficult to achieve. This is partly solved by varying the quantization (quality), and fully solved by the use of padding. (However, CBR is implied in a simple scheme like reducing all 16-bit audio samples to 8 bits.)





Variable bitrate (VBR) is a term used in telecommunications and computing that relates to the bitrate used in sound or video encoding. As opposed to constant bitrate (CBR), VBR files vary the amount of output data per time segment. VBR allows a higher bitrate (and therefore more storage space) to be allocated to the more complex segments of media files while less space is allocated to less complex segments. The average of these rates can be calculated to produce an average bitrate for the file.The advantages of VBR are that it produces a better quality-to-space ratio compared to a CBR file of the same data. The bits available are used more flexibly to encode the sound or video data more accurately, with fewer bits used in less demanding passages and more bits used in difficult-to-encode passages.This VBR encoding method allows the user to specify a bitrate range - a minimum and/or maximum allowed bitrate.[15] Some encoders extend this method with an average bitrate. The minimum and maximum allowed bitrate set bounds in which the bitrate may vary. The disadvantage of this method is that the average bitrate (and hence file size) will not be known ahead of time. The bitrate range is also used in some fixed quality encoding methods, but usually without permission to change a particular bitrate





(iii) Time Division Multiplexing and Frequency Division Multiplexing

ans : Time-division multiplexing (TDM) is a type of digital or (rarely) analog multiplexing in which two or more signals or bit streams are transferred apparently simultaneously as sub-channels in one communication channel, but are physically taking turns on the channel. The time domain is divided into several recurrent timeslots of fixed length, one for each sub-channel. A sample byte or data block of sub-channel 1 is transmitted during timeslot 1, sub-channel 2 during timeslot 2, etc. One TDM frame consists of one timeslot per sub-channel plus a synchronization channel and sometimes error correction channel before the synchronization. After the last sub-channel, error correction, and synchronization, the cycle starts all over again with a new frame, starting with the second sample, byte or data block from sub-channel 1, etc.



TDM is used for circuit mode communication with a fixed number of channels and constant bandwidth per channel.



Bandwidth Reservation distinguishes time-division multiplexing from statistical multiplexing such as packet mode communication (also known as statistical time-domain multiplexing, see below) i.e. the time-slots are recurrent in a fixed order and pre-allocated to the channels, rather than scheduled on a packet-by-packet basis. Statistical time-domain multiplexing resembles, but should not be considered the same as time-division multiplexing.



Frequency-division multiplexing (FDM) is a form of signal multiplexing which involves assigning non-overlapping frequency ranges to different signals or to each "user" of a medium.FDM can also be used to combine signals before final modulation onto a carrier wave. In this case the carrier signals are referred to as subcarriers: an example is stereo FM transmission, where a 38 kHz subcarrier is used to separate the left-right difference signal from the central left-right sum channel, prior to the frequency modulation of the composite signal. A television channel is divided into subcarrier frequencies for video, color, and audio. DSL uses different frequencies for voice and for upstream and downstream data transmission on the same conductors, which is also an example of frequency duplex.



Where frequency-division multiplexing is used as to allow multiple users to share a physical communications channel, it is called frequency-division multiple access (FDMA).[1]



FDMA is the traditional way of separating radio signals from different transmitters.



In the 1860s and 70s, several inventors attempted FDM under the names of Acoustic telegraphy and Harmonic telegraphy. Practical FDM was only achieved in the electronic age. Meanwhile their efforts led to an elementary understanding of electroacoustic technology, resulting in the invention of the telephone.For long distance telephone connections, 20th century telephone companies used L-carrier and similar co-axial cable systems carrying thousands of voice circuits multiplexed in multiple stages by channel banks.



For shorter distances,cheaper balanced pair cables were used for various systems including Bell System K- and N-Carrier. Those cables didn't allow such large bandwidths, so only 12 voice channels (Double Sideband) and later 24 (Single Sideband) were multiplexed into four wires, one pair for each direction with repeaters every several miles, approximately 10 km. See 12-channel carrier system. By the end of the 20th Century, FDM voice circuits had become rare. Modern telephone systems employ digital transmission, in which time-division multiplexing (TDM) is used instead of FDM.



Since the late 20th century Digital Subscriber Lines have used a Discrete multitone (DMT) system to divide their spectrum into frequency channels.



The concept corresponding to frequency-division multiplexing in the optical domain is known as wavelength division multiplexing.



(iv) Datagram and Virtual Circuit

ans : Virtual circuit packet switching sets up a single path along which all packets in the message will travel. The facilities along that path are not dedicated to the circuit and may be used by other packets as well as those traveling through the virtual circuit. (For circuit switching the path is dedicated so traffic that is bursty wastes capacity in a circuit switched network but not a virtual circuit packet switching network)

The setup of the circuit is one source of overhead that is not present in datagram packet switching. Datagram packet switching sends each packet along the path that is optimal at the time the packet is sent. When a packet traverses the network each intermediate station will need to determine the next hop. This should be equally efficient in both virtual circuit packet switching and datagram packet switching. Virtual circuit packet switching must find the entry for the flow in the routing table, datagram packet switching must find the entry for the destination in the routing table.

For datagram packet switching in a real network the path of each packet is determined independently. Each packet may travel by a different path. Each different path will have a different total transmission delay (the number of hops in the path may be different, and the delay across each hop may change for different routes). Therefore, it is possible for the packets to arrive at the destination in a different order from the order in which they were sent. In contrast, for virtual circuit packet switching, all packets follow the same path, through the same virtual circuit. Therefore, for virtual circuit packet switching the packet will arrive in the order they are sent. In a congested network it is possible that virtual circuit packet switching will experience additional queuing delays at busy intermediate stations, while datagram packet switching may be able to avoid that congestion and delay by choosing other paths for the packets.

Packet size and packet transmission time (the time to insert the packet into the network) are important in both types of packet switching. The packet headers used in both approaches will be the same size given that the protocols used in both networks are the same. (Source routing, where all intermediate stations are listed in the header is an exception to this rule). There is an optimal packet size where the tradeoff between extra overhead due to packet headers and shorter packets balances. Smaller packets lead to more equitable sharing of facilities between processes. Each process will have a shorter wait for its ‘turn’ to transmit. A smaller packet is also less likely to contain an error and need to be discarded or retransmitted. A smaller packet also means there is less data lost or retransmitted when an error does occur. These improvements in efficiency must be balanced against the added overhead, smaller packets mean more packets to transmit the same amount of data. Each packet has a header, so more packets mean more capacity used by the overhead of transmitting packet headers.

The delay caused by waiting for packets to arrive at intermediate stations has two sources, the transmission (or equivalent reception) time and any time the packets sit it a local queue waiting to be transmitted. The first source of delay is the same for virtual circuit packet switching and datagram packet switching given that the packets for each method are the same size. The latter depends on present conditions in the network and under different condition may favor either method.



(v) Analog and Digital Signal



Ans : Analog signals are continuous where digital signals are discrete. Anolog signals are continuously varying where digital signals are based on 0's and 1's (or as often said------- on's and off's). As an analogy, consider a light switch that is either on or off (digital) and a dimmer switch (analog) that allows you to vary the light in different degrees of brightness. As another analogy, consider a clock in which the second hand smoothly circles the clock face (analog) versus another clock in which the second hand jumps as each second passes (digital). Digital computers work with a series of 0's and 1's to represent letters, symbols, and numbers. In addition, numbers are represented by using the binary code (where only 0's and 1's are used).



Number Binary equivalent

1----------------------------------------------1

2---------------------------------------------10

3---------------------------------------------11

4--------------------------------------------100

5--------------------------------------------101

6--------------------------------------------110

7--------------------------------------------111

8-------------------------------------------1000



and so on. So each number (that we are accustomed to, such as 5) is represented by 0's and 1's. Morse code uses dits (or dots) and dashes. Digital signals are similar to Morse code. The signal is either a dit or a dash for Morse code and it is either a 0 or 1 for digital. A series of these dits and dashes might represent SOS to a navy radio man, and a series of 0's and 1's might represent the question mark to a computer.



When an e-mail is sent that says "Hello Joe", Hello Joe doesn't mysteriously appear on Joe's computer. What is sent through the phone line is a series of 0's and 1's and Joe's computer "interprets" these into the words Hello Joe. If you type the letter A into your computer, it converts this A into 01000001. This 01000001 goes to Joe's computer and his computer interprets it as A. Each 0 or 1 is "bit" and the series of eight 0's and 1's is a byte. Well, that is about as simple as it gets and about as simple as I can state it.



Analog Signal:

1.Analog signal are continuous.

2.Analog signal is continuously variable.

3.The primary disadvantage of an analog signal is noise.

4.Sound waves are a continuous wave and as such are

analog in the real world.

5.Analog signal required lesser bandwidth capacity than digital capacity.



Digital Signal:

1.Digital signal is discrete.

2.Digital signal are based on 0's and 1's.

3.Noise is much easier to filter out of a digital signal.

4.Most computer used such as the PC work using digital signals.

5.Digital signal required greater bandwidth capacity than analog signals.



An analog signal is continuously variable. It differs from a digital signal in that small fluctuations in the signal are meaningful. That is the key. Whereas a digital signal only represents 2 values (0 and 1 - or off and on).



The primary disadvantage of an analog signal is noise. As an analog signal is processed (copied, sampled, amplified, etc) the noise is hard to discriminate from the actual signal. Noise is much easier to filter out of a digital signal, because anything other than the pure 'high' or 'low' signal is considered noise.



In an analog signal the voltage may assume any numeric value within some continuous range, changing smoothly with time.



In a digital signal the voltage may assume only two discrete values, jumping between one & the other with time. These are square pulses.



A pictorial diagram would be far easier to understand than this verbal description, but that's the best we can do here. Hope this helps.





Question2: (i) Explain through examples the followings in terms of TCP reference model.

*Reliable connection–oriented transfer of data

*Best effort connectionless transfer



(ii) Explain with illustration how the Internet Layer and the network Interface layer of TCP reference model function.



(iii) List advantages of digital transmission over analog transmission

with examples of each.



Ans :

(i)TCP reference model for given terms are as follows : -



Reliable connection–oriented transfer of data



connection-oriented communication is a data communication mode in which the devices at the end points use a protocol to establish an end-to-end logical or physical connection before any data may be sent. In case of digital transmission, in-order delivery of a bit stream or byte stream is provided. Connection-oriented protocol services are often but not always reliable network services, that provide acknowledgment after successful delivery, and automatic repeat request functions in case of missing data or detected bit-errors.



Circuit mode communication, for example the public switched telephone network, ISDN, SONET/SDH and optical mesh networks, are examples of connection-oriented communication. Circuit mode communication provides guarantees that data will arrive with constant bandwidth and at constant delay.



Packet mode communication may also be connection-oriented, which is called virtual circuit mode communication. Due to the packet switching, the communication may suffer from variable bit rate and delay, due to varying traffic load and packet queue lengths.



A connection-oriented transport layer protocol, such as TCP, may be based on a connectionless network layer protocol (such as IP), but still achieve in-order delivery of a byte-stream, by means of segment numbering on the sender side and data packet reordering on the receiver side.



In a connection-oriented packet switched data link layer or network layer protocol, all data is sent over the same path during a communication session. The protocol does not have to provide each packet with routing information (complete source and destination address), but only with a channel/data stream number, often denoted virtual circuit identifier (VCI). Routing information may be provided to the network nodes during the connection establishment phase, where the VCI is defined in tables in each node. Thus, the actual packet switching and data transfer can be taken care of by fast hardware, as opposed to slow software based routing.





Examples of connection-oriented packet mode communication, i.e. virtual circuit mode communication:



* The Transmission Control Protocol (TCP) is a connection-oriented reliable protocol that is based on a datagram protocol (the IP protocol).

* X.25 is a connection-oriented reliable network protocol.

* Frame relay is a connection-oriented unreliable data link layer protocol.

* GPRS

* Asynchronous Transfer Mode

* Multiprotocol Label Switching



Best effort connectionless transfer



Implements two functions: addressing and fragmentation.

• IP encapsulates data handed to it from its upper-layer software with its headers.

• IP delivers data based on a best effort.

– Transmits an encapsulated packet and does not expect a response

• IP receives data handed to it by the datalink.

– Decapsulates a packet (strips its headers off) and hands the data to its upper-layer software

The IP layer provides the entry into the delivery system used to transport data across the Internet. Usually, when anyone hears the name IP, he or she automatically thinks of the networks connected together through devices commonly known as routers, which connect multiple subnetworks together. It is true the IP performs these tasks, but the IP protocol performs many other tasks, as mentioned previously. The IP protocol runs in all the participating network stations that are attached to subnetworks so that they may submit their packets to routers or directly to other devices on the same network. It resides between the datalink layer and the transport layer. IP also provides for connectionless data delivery between nodes on an IP network.

The primary goal of IP is to provide the basic algorithm for transfer of data to and from a network. In order to achieve this, it implements two functions: addressing and fragmentation. It provides a connectionless delivery service for the upper-layer protocols. This means that IP does not set up a session (a virtual link) between the transmitting station and the receiving station prior to submitting the data to the receiving station. It encapsulates the data handed to it and delivers it on a best-effort basis. IP does not inform the sender or receiver of the status of the packet; it merely attempts to deliver the packet and will not make up for the faults encountered in this attempt. This means that if the datalink fails or incurs a recoverable error, the IP layer will not inform anyone. It tried to deliver (addressed) a message and failed. It is up to the upper-layer protocols (TCP, or even the application itself) to perform error recovery. For example, if your application is using TCP as its transport layer protocol, TCP will time-out for that transmission and will resend the data. If the application is using UDP as its transport, then it is up to the application to perform error recovery procedures.

IP submits a properly formatted data packet to the destination station and does not expect a status response. Because IP is a connectionless protocol, IP may receive and deliver the data (data sent to the transport layer in the receiving station) in the wrong order from which it was sent, or it may duplicate the data. Again, it is up to the higher-layer protocols (layer 4 and above) to provide error recovery procedures. IP is part of the network delivery system. It accepts data and formats it for transmission to the datalink layer. (Remember, the datalink layer provides the access methods to transmit and receive data from the attached cable plant.) IP also retrieves data from the datalink and presents it to the requesting upper layer.







(ii) The Internet Layer is a group of internetworking methods in the Internet Protocol Suite, commonly also called TCP/IP, which is the foundation of the Internet. It is the group of methods, protocols, and specifications that are used to transport datagrams (packets) from the originating host across network boundaries, if necessary, to the destination host specified by a network address (IP address) which is defined for this purpose by the Internet Protocol (IP). The Internet Layer derives its name from its function of forming an internet (uncapitalized), or facilitating internetworking, which is the concept of connecting multiple networks with each other through gateways.



Internet Layer protocols use IP-based packets. The Internet Layer does not include the protocols that define communication between local (on-link) network nodes which fulfill the purpose of maintaining link states between the local nodes, such as the local network topology, and that usually use protocols that are based on the framing of packets specific to the link types. Such protocols belong to the Link Layer.



A particularly crucial aspect in the Internet Layer is the Robustness Principle: "Be liberal in what you accept, and conservative in what you send" (RFC 1122), as a misbehaving host can deny Internet service to many other users.

The Internet Layer has three basic functions: For outgoing packets, select the next-hop host (gateway) and transmit the packet to this host by passing it to the appropriate Link Layer implementation; for incoming packets, capture packets and pass the packet payload up to the appropriate Transport Layer protocol, if appropriate. In addition it provides error detection and diagnostic capability.



In Version 4 of the Internet Protocol (IPv4), during both transmit and receive operations, IP is capable of automatic or intentional fragmentation or defragmentation of packets, based, for example, on the maximum transmission unit (MTU) of link elements. However, this feature has been dropped in IPv6, as the communications end points, the hosts, now have to perform path MTU discovery and assure that end-to-end transmissions don't exceed the maximum discovered.



In its operation, the Internet Layer is not responsible for reliable transmission. It provides only an unreliable service, and "best effort" delivery. This means that the network makes no guarantees about packets' proper arrival (see also Internet Protocol#Reliability). This was an important design principle and change from the previous protocols used on the early ARPANET. Since packet delivery across diverse networks is inherently an unreliable and failure-prone operation, the burden of providing reliability was placed with the end points of a communication path, i.e., the hosts, rather than on the network. This is one of the reasons of the resiliency of the Internet against individual link failures and its proven scalability.



The function of providing reliability of service is the duty of higher level protocols, such as the Transmission Control Protocol (TCP) in the Transport Layer.



Integrity of packets is guaranteed only in IPv4 (not in IPv6) through checksums computed for IP packets.





The Network Layer is Layer 3 of the seven-layer OSI model of computer networking.



The Network Layer is responsible for routing packets delivery including routing through intermediate routers, whereas the Data Link Layer is responsible for Media Access Control, Flow Control and Error Checking.



The Network Layer provides the functional and procedural means of transferring variable length data sequences from a source to a destination host via one or more networks while maintaining the quality of service functions.



Functions of the Network Layer include:



* Connection model: connectionless communication



For example, IP is connectionless, in that a frame can travel from a sender to a recipient without the recipient having to send an acknowledgement. Connection-oriented protocols exist at other higher layers of that model.



* Host addressing



Every host in the network needs to have a unique address which determines where it is. This address will normally be assigned from a hierarchical system, so you can be "Fred Murphy" to people in your house, "Fred Murphy, Main Street 1" to Dubliners, or "Fred Murphy, Main Street 1, Dublin" to people in Ireland, or "Fred Murphy, Main Street 1, Dublin, Ireland" to people anywhere in the world. On the Internet, addresses are known as Internet Protocol (IP) addresses.



* Message forwarding



Since many networks are partitioned into subnetworks and connect to other networks for wide-area communications, networks use specialized hosts, called gateways or routers to forward packets between networks. This is also of interest to mobile applications, where a user may move from one location to another, and it must be arranged that his messages follow him. Version 4 of the Internet Protocol (IPv4) was not designed with this feature in mind, although mobility extensions exist. IPv6 has a better designed solution.



Within the service layering semantics of the OSI network architecture the Network Layer responds to service requests from the Transport Layer and issues service requests to the Data Link Layer.



(iii) Advantages of digital transmission over analog transmission :-



Analog vs. Digital Transmission

Compare at two levels:

1. Data|continuous (audio) vs. discrete (text)

2. Signaling|continuously varying electromagnetic wave vs. sequence of voltage pulses.

Also Transmission|transmit without regard to signal content vs. being concerned with

signal content. Di

erence in how attenuation is handled, but not focus on this.

Look at Table 2.3 and Fig 2.13 in Stallings.

Seeing a shift towards digital transmission despite large analog base. Why?

improving digital technology

data integrity. Repeaters take out cumulative problems in transmission. Can thus

transmit longer distances.

easier to multiplex large channel capacities with digital

easy to apply encryption to digital data

better integration if all signals are in one form. Can integrate voice, video and digital

data.



Digital Data/Analog Signals



Must convert digital data to analog signal. One such device is a modem to translate

between bit-serial and modulated carrier signals.

To send digital data using analog technology, the sender generates a carrier signal at some

continuous tone (e.g. 1-2 kHz in phone circuits) that looks like a sine wave. The following

techniques are used to encode digital data into analog signals (Fig 2-18)

Resulting bandwidth is centered on the carrier frequency.

amplitude-shift modulation (keying): vary the amplitude (e.g. voltage) of the signal.

Used to transmit digital data over optical ber.

frequency-shift modulation: two (or more tones) are used, which are near the carrier

frequency. Used in a full-duplex modem (signals in both directions).

phase-shift modulation: systematically shift the carrier wave at uniformly spaced

intervals.

For instance, the wave could be shifted by 45, 135, 225, 315 degree at each timing

mark. In this case, each timing interval carries 2 bits of information.

Why not shift by 0, 90, 180, 270? Shifting zero degrees means no shift, and an

extended set of no shifts leads to clock synchronization diculties.

Another variation, called Quadrature Amplitude Modulation, has the following

characteristics:

Look at Figure 2-25. Can use:

QAM. 4 phase shifts|2-bit encoding. (Quadrature Phase-Shift Keying)

QAM-16. 4 phase shifts plus four amplitudes|4-bit encoding.

Used with 2400 baud we get an e

ective data rate of 9600bps.

QAM-64. 6-bit encoding.

V.32 for 9600 bps and combine with error correction (Fig 2-26).

V.34 12-bit encoding



Question 3: (i) Describe the Selective Repeat Protocol with illustration in terms of

error recovery , flow control and packet loss.

(ii) How is MAC sublayer different from Logical Link Control sublayer?

(iii) Describe the two methodologies used by bridges for transmission of

frames.

(iv) What function(s) a station has to perform when it wants to transmit

a frame using a token ring protocol?





Ans : (i)



Selective Repeat ARQ / Selective Reject ARQ is a specific instance of the Automatic Repeat-reQuest (ARQ) Protocol. It may be used as a protocol for the delivery and acknowledgement of message units, or it may be used as a protocol for the delivery of subdivided message sub-units.



When used as the protocol for the delivery of messages, the sending process continues to send a number of frames specified by a window size even after a frame loss. Unlike Go-Back-N ARQ, the receiving process will continue to accept and acknowledge frames sent after an initial error; this is the general case of the sliding window protocol with both transmit and receive window sizes greater than 1.



The receiver process keeps track of the sequence number of the earliest frame it has not received, and sends that number with every acknowledgement (ACK) it sends. If a frame from the sender does not reach the receiver, the sender continues to send subsequent frames until it has emptied its window. The receiver continues to fill its receiving window with the subsequent frames, replying each time with an ACK containing the sequence number of the earliest missing frame. Once the sender has sent all the frames in its window, it re-sends the frame number given by the ACKs, and then continues where it left off.



The size of the sending and receiving windows must be equal, and half the maximum sequence number (assuming that sequence numbers are numbered from 0 to n−1) to avoid miscommunication in all cases of packets being dropped. To understand this, consider the case when all ACKs are destroyed. If the receiving window is larger than half the maximum sequence number, some, possibly even all, of the packages that are resent after timeouts are duplicates that are not recognized as such. The sender moves its window for every packet that is acknowledged.

Selective Repeat is a connection oriented protocol in which both transmitter and receiver have a window of sequence numbers.



The protocol simulation shows a time-sequence diagram with users A and B, protocol entities A and B that support them, and a communications medium that carries messages. Users request data transmissions with DatReq(DATAn), and receive data transmissions as DatInd(DATAn). Data messages are simply numbered DATA0, DATA1, etc. without explicit content. The transmitting protocol sends the protocol message DT(n) that gives only the sequence number, not the data. Once sequence numbers reach a maximum number (like 7), they wrap back round to 0. An acknowledgement AK(n) means that the DT message numbered n is the next one expected (i.e. all messages up to but not including this number have been received). Since sequence numbers wrap round, an acknowledgement with sequence number 1 refers to messages 0, 1, 7, 6, etc. Note that if a DT message is received again due to re-transmission, it is acknowledged but discarded.



The protocol has a maximum number of messages that can be sent without acknowledgement. If this window becomes full, the protocol is blocked until an acknowledgement is received for the earliest outstanding message. At this point the transmitter is clear to send more messages.



The receiver delivers the protocol messages DT(n) to the user in order. Any received out of order, but within the receiver's window are buffered.



(ii)

Mac Layer

The Media Access Control (MAC) data communication protocol sub-layer, also known as the Medium Access Control, is a sublayer of the Data Link Layer specified in the seven-layer OSI model (layer 2). It provides addressing and channel access control mechanisms that make it possible for several terminals or network nodes to communicate within a multi-point network, typically a local area network (LAN) or metropolitan area network (MAN). The hardware that implements the MAC is referred to as a Medium Access Controller.



The MAC sub-layer acts as an interface between the Logical Link Control (LLC) sublayer and the network's physical layer. The MAC layer emulates a full-duplex logical communication channel in a multi-point network. This channel may provide unicast, multicast or broadcast communication service.

A MAC address is a unique serial number. Once a MAC address has been assigned to a particular network interface (typically at time of manufacture), that device should be uniquely identifiable amongst all other network devices in the world. This guarantees that each device in a network will have a different MAC address (analogous to a street address). This makes it possible for data packets to be delivered to a destination within a subnetwork, i.e. a physical network consisting of several network segments interconnected by repeaters, hubs, bridges and switches, but not by IP routers. An IP router may interconnect several subnets.



LLC Layer

The Logical Link Control (LLC) data communication protocol layer is the upper sub-layer of the Data Link Layer (which is itself layer 2, just above the Physical Layer) in the seven-layer OSI reference model. It provides multiplexing mechanisms that make it possible for several network protocols (IP, IPX) to coexist within a multipoint network and to be transported over the same network media, and can also provide flow control mechanisms.



The LLC sub-layer acts as an interface between the Media Access Control (MAC) sublayer and the network layer.



As the Ethertype in an Ethernet II framing formatted frame is used to multiplex different protocols on top of the Ethernet MAC header it can be seen as LLC identifier.



Some non-IEEE 802 protocols can be thought of as being split into MAC and LLC layers. For example, while HDLC specifies both MAC functions (framing of packets) and LLC functions (protocol multiplexing, flow control, detection, and error control through a retransmission of dropped packets when indicated), some protocols such as Cisco HDLC can use HDLC-like packet framing and their own LLC protocol.



An LLC header tells the Data Link layer what to do with a packet once a frame is received. It works like this: A host will receive a frame and look in the LLC header to find out where the packet is destined for - for example, the IP protocol at the Network layer or IPX.



(iii)

1.A data transmission installation for forwarding data transmission frames, the data transmission installation comprising data receivers for receiving data transmission frames in the data transmission installation, data transmitters for sending data transmission frames from the data transmission installation, and a processor unit adapted to:read control data from one or more received data transmission frames, the control data containing building and cancelling commands for logical data transmission tunnels determined on the basis of mobility of data terminals connected to a data transmission network,execute building and cancelling actions determined by the control data for logical data transmission tunnels, anddetermine forwarding actions for a received second data transmission frame on the basis of forwarding control data contained in the second data transmission frame, the forwarding actions comprising a transmission of the second data transmission frame by way of one of following (i-iii): (i) a predetermined output port of the data transmission installation or (ii) one of two optional predetermined output ports of the data transmission installation or (iii) a logical data transmission tunnel commencing from the data transmission installation.



2. A data transmission installation according to claim 1, wherein the processor unit is adapted to read second control data from one or more received third data transmission frames and to select one of the two optional predetermined output ports on the basis of the second control data and forwarding control data contained in the second data transmission frame.



(iv)



Token ring local area network (LAN) technology is a local area network protocol which resides at the data link layer (DLL) of the OSI model. It uses a special three-byte frame called a token that travels around the ring. Token-possession grants the possessor permission to transmit on the medium. Token ring frames travel completely around the loop.When no station is transmitting a data frame, a special token frame circles the loop. This special token frame is repeated from station to station until arriving at a station that needs to transmit data. When a station needs to transmit data, it converts the token frame into a data frame for transmission. Once the sending station receives its own data frame, it converts the frame back into a token. If a transmission error occurs and no token frame, or more than one, is present, a special station referred to as the Active Monitor detects the problem and removes and/or reinserts tokens as necessary (see Active and standby monitors). On 4 Mbit/s Token Ring, only one token may circulate; on 16 Mbit/s Token Ring, there may be multiple tokens.





Question 4: (i) How does the transport layer manage the crash recovery,

multiplexing and connection establishment? Explain

(ii) What is the convention of writing domain names? Explain through

an example. List at least five domain names and its meaning.



Ans : (i)



In computer networking, the Transport Layer provides end-to-end communication services for applications[1] within a layered architecture of network components and protocols. The transport layer provides convenient services such as connection-oriented data stream support, reliability, flow control, and multiplexing.



Transport layers are contained in both the TCP/IP model (RFC 1122),[2] which is the foundation of the Internet, and the Open Systems Interconnection (OSI) model of general networking. The definitions of the Transport Layer are slightly different in these two models. This article primarily refers to the TCP/IP model, in which TCP is largely for a convenient application programming interface to internet hosts, as opposed to the OSI model definition of the Transport Layer.



The most well-known transport protocol is the Transmission Control Protocol (TCP). It lent its name to the title of the entire Internet Protocol Suite, TCP/IP. It is used for connection-oriented transmissions, whereas the connectionless User Datagram Protocol (UDP) is used for simpler messaging transmissions. TCP is the more complex protocol, due to its stateful design incorporating reliable transmission and data stream services. Other prominent protocols in this group are the Datagram Congestion Control Protocol (DCCP) and the Stream Control Transmission Protocol (SCTP).



The Transport Layer is responsible for delivering data to the appropriate application process on the host computers. This involves statistical multiplexing of data from different application processes, i.e. forming data packets, and adding source and destination port numbers in the header of each Transport Layer data packet. Together with the source and destination IP address, the port numbers constitutes a network socket, i.e. an identification address of the process-to-process communication. In the OSI model, this function is supported by the Session Layer.



Some Transport Layer protocols, for example TCP, but not UDP, support virtual circuits, i.e. provide connection oriented communication over an underlying packet oriented datagram network. A byte-stream is delivered while hiding the packet mode communication for the application processes. This involves connection establishment, dividing of the data stream into packets called segments, segment numbering and reordering of out-of order data.



Finally, some Transport Layer protocols, for example TCP, but not UDP, provide end-to-end reliable communication, i.e. error recovery by means of error detecting code and automatic repeat request (ARQ) protocol. The ARQ protocol also provides flow control, which may be combined with congestion avoidance.



UDP is a very simple protocol, and does not provide virtual circuits, nor reliable communication, delegating these functions to the application program. UDP packets are called datagrams, rather than segments.



TCP is used for many protocols, including HTTP web browsing and email transfer. UDP may be used for multicasting and broadcasting, since retransmissions are not possible to a large amount of hosts. UDP typically gives higher throughput and shorter latency, and is therefore often used for real-time multimedia communication where packet loss occasionally can be accepted, for example IP-TV and IP-telephony, and for online computer games.



In many non-IP-based networks, for example X.25, Frame Relay and ATM, the connection oriented communication is implemented at network layer or data link layer rather than the Transport Layer. In X.25, in telephone network modems and in wireless communication systems, reliable node-to-node communication is implemented at lower protocol layers.



There are many services that can be optionally provided by a Transport Layer protocol, and different protocols may or may not implement them.



* Connection-oriented communication: Interpreting the connection as a data stream can provide many benefits to applications. It is normally easier to deal with than the underlying connection-less models, such as the Transmission Control Protocol's underlying Internet Protocol model of datagrams.

* Byte orientation: Rather than processing the messages in the underlying communication system format, it is often easier for an application to process the data stream as a sequence of bytes. This simplification helps applications work with various underlying message formats.

* Same order delivery: The Network layer doesn't generally guarantee that packets of data will arrive in the same order that they were sent, but often this is a desirable feature. This is usually done through the use of segment numbering, with the receiver passing them to the application in order. This can cause head-of-line blocking.

* Reliability: Packets may be lost during transport due to network congestion and errors. By means of an error detection code, such as a checksum, the transport protocol may check that the data is not corrupted, and verify correct receipt by sending an ACK or NACK message to the sender. Automatic repeat request schemes may be used to retransmit lost or corrupted data.

* Flow control: The rate of data transmission between two nodes must sometimes be managed to prevent a fast sender from transmitting more data than can be supported by the receiving data buffer, causing a buffer overrun. This can also be used to improve efficiency by reducing buffer underrun.

* Congestion avoidance: Congestion control can control traffic entry into a telecommunications network, so as to avoid congestive collapse by attempting to avoid oversubscription of any of the processing or link capabilities of the intermediate nodes and networks and taking resource reducing steps, such as reducing the rate of sending packets. For example, automatic repeat requests may keep the network in a congested state; this situation can be avoided by adding congestion avoidance to the flow control, including slow-start. This keeps the bandwidth consumption at a low level in the beginning of the transmission, or after packet retransmission.

* Multiplexing: Ports can provide multiple endpoints on a single node. For example, the name on a postal address is a kind of multiplexing, and distinguishes between different recipients of the same location. Computer applications will each listen for information on their own ports, which enables the use of more than one network service at the same time. It is part of the Transport Layer in the TCP/IP model, but of the Session Layer in the OSI model.



(ii)





A domain name is an identification label that defines a realm of administrative autonomy, authority, or control in the Internet. Domain names are also hostnames that identify Internet Protocol (IP) resources such as web sites. Domain names are formed by the rules and procedures of the Domain Name System (DNS).



Domain names are used in various networking contexts and application-specific naming and addressing purposes. They are organized in subordinate levels (subdomains) of the DNS root domain, which is nameless. The first-level set of domain names are the top-level domains (TLDs), including the generic top-level domains (gTLDs), such as the prominent domains com, net and org, and the country code top-level domains (ccTLDs). Below these top-level domains in the DNS hierarchy are the second-level and third-level domain names that are typically open for reservation by end-users that wish to connect local area networks to the Internet, run web sites, or create other publicly accessible Internet resources. The registration of these domain names is usually administered by domain name registrars who sell their services to the public.



Individual Internet host computers use domain names as host identifiers, or hostnames. Hostnames are the leaf labels in the domain name system usually without further subordinate domain name space. Hostnames appear as a component in Uniform Resource Locators (URLs) for Internet resources such as web sites (e.g., en.wikipedia.org).



Domain names are also used as simple identification labels to indicate ownership or control of a resource. Such examples are the realm identifiers used in the Session Initiation Protocol (SIP), the DomainKeys used to verify DNS domains in e-mail systems, and in many other Uniform Resource Identifiers (URIs).



An important purpose of domain names is to provide easily recognizable and memorizable names to numerically addressed Internet resources. This abstraction allows any resource (e.g., website) to be moved to a different physical location in the address topology of the network, globally or locally in an intranet. Such a move usually requires changing the IP address of a resource and the corresponding translation of this IP address to and from its domain name.



Domain names are often referred to simply as domains and domain name registrants are frequently referred to as domain owners, although domain name registration with a registrar does not confer any legal ownership of the domain name, only an exclusive right of use.



The Internet Corporation for Assigned Names and Numbers (ICANN) manages the top-level development and architecture of the Internet domain name space. It authorizes domain name registrars, through which domain names may be registered and reassigned. The use of domain names in commerce may subject strings in them to trademark law. In 2010, the number of active domains reached 196 million.[

A domain name is an identification label that defines a realm of administrative autonomy, authority, or control in the Internet. Domain names are also hostnames that identify Internet Protocol (IP) resources such as web sites. Domain names are formed by the rules and procedures of the Domain Name System (DNS).



Domain names are used in various networking contexts and application-specific naming and addressing purposes. They are organized in subordinate levels (subdomains) of the DNS root domain, which is nameless. The first-level set of domain names are the top-level domains (TLDs), including the generic top-level domains (gTLDs), such as the prominent domains com, net and org, and the country code top-level domains (ccTLDs). Below these top-level domains in the DNS hierarchy are the second-level and third-level domain names that are typically open for reservation by end-users that wish to connect local area networks to the Internet, run web sites, or create other publicly accessible Internet resources. The registration of these domain names is usually administered by domain name registrars who sell their services to the public.



Individual Internet host computers use domain names as host identifiers, or hostnames. Hostnames are the leaf labels in the domain name system usually without further subordinate domain name space. Hostnames appear as a component in Uniform Resource Locators (URLs) for Internet resources such as web sites (e.g., en.wikipedia.org).



Domain names are also used as simple identification labels to indicate ownership or control of a resource. Such examples are the realm identifiers used in the Session Initiation Protocol (SIP), the DomainKeys used to verify DNS domains in e-mail systems, and in many other Uniform Resource Identifiers (URIs).



An important purpose of domain names is to provide easily recognizable and memorizable names to numerically addressed Internet resources. This abstraction allows any resource (e.g., website) to be moved to a different physical location in the address topology of the network, globally or locally in an intranet. Such a move usually requires changing the IP address of a resource and the corresponding translation of this IP address to and from its domain name.



Domain names are often referred to simply as domains and domain name registrants are frequently referred to as domain owners, although domain name registration with a registrar does not confer any legal ownership of the domain name, only an exclusive right of use.



The Internet Corporation for Assigned Names and Numbers (ICANN) manages the top-level development and architecture of the Internet domain name space. It authorizes domain name registrars, through which domain names may be registered and reassigned. The use of domain names in commerce may subject strings in them to trademark law. In 2010, the number of active domains reached 196 million.



Five domain names and its meaning : -



.net represents the word “network,” and is most commonly

used by Internet service providers, Web-hosting companies or other businesses

that are directly involved in the infrastructure of the Internet. Additionally,

some businesses choose domain names with a .net extension for their intranet

Websites.



.org represents the word “organization” and is primarily

used by non-profits groups or trade associations.



.biz is used for small business web sites.



.info is for credible resource web sites and signifies a

“resource” web site. It’s the most popular extension beyond .com, .net and .org.



.mobi (short for “mobile”) is reserved for Web sites built

for easy viewing on mobile devices.



.us is for American Web sites and is the newest extension.

It has the largest amount of available names in inventory.





Question 5: (i) Through an example and an illustration, explain how an ATM network work?

(ii) Describe the bandwidth limitations of B-channels and D-channel.

(iii) Differentiate between Packet switching and Cell switching.



Ans : (i)



Asynchronous Transfer Mode (ATM) is a switching technique for telecommunication networks. It uses asynchronous time-division multiplexing,[1][2] and it encodes data into small, fixed-sized cells. This differs from networks such as the Internet or Ethernet LANs that use variable sized packets or frames. ATM provides data link layer services that run over OSI Layer 1 physical links. ATM has functional similarity with both circuit switched networking and small packet switched networking. This makes it a good choice for a network that must handle both traditional high-speed data traffic (e.g., file transfers), and real-time, low-latency content such as voice and video. ATM uses a connection-oriented model in which a virtual circuit must be established between two endpoints before the actual data exchange begins.[2] ATM is a core protocol used over the SONET/SDH backbone of the Integrated Services Digital Network (ISDN).



ATM Network Element (ATM NE-1) enabling automatic protection switching of a Transmission Convergence Sublayer SubNetwork Connection and comprising at least one working Transmission Convergence Sublayer entity (TCS-W) as part of a first physical line interface and at least one protection Transmission Convergence Sublayer entity (TCS-P) as part of a second physical line interface, both entities (TCS-W,TCS-P), respectively, being individually addressable within the ATM Network Element (ATM NE-1) via an assigned UTOPIA address, the ATM Network Element (ATM NE-1) being provided by the Transmission Convergence Sublayer with physical layer error messages indicating failures in the Transmission Convergence Sublayer SubNetwork Connection established via the working Transmission Convergence Sublayer entity (TCS-W), the ATM Network Element (ATM NE-1) being configured to select for transmission of ATM-cells the UTOPIA address (UA1) of the working Transmission Convergence Sublayer entity (TCS-W) as long as no physical layer error message is received, and to switch the selection, after reception of a physical layer error message, to the UTOPIA address (UA2) of the protection Transmission Convergence Sublayer entity (TCS-P), thereby providing automatic protection switching of the Transmission Convergence Sublayer SubNetwork Connection without requiring an action in an ATM layer of the ATM network element (ATM NE-1) and the ATM Network Element (ATM NE-1) being configured to merge the data entering the ATM Network Element (ATM NE-1) via both, the working Transmission Convergence Sublayer entity (TCS-W) and the protection Transmission Convergence Sublayer entity (TCS-P).



2. ATM Network Element (ATM NE-1) according to claim 1, wherein the physical line interfaces are SDH and/or PDH physical line interfaces.



3. ATM Network Element (ATM NE-1) according to claim 2, wherein the physical line interfaces are PDH physical line interfaces used for inverse multiplexing for ATM (IMA).



4. ATM Network Element (ATM NE-1) according to claim 1, wherein the physical layer error messages are OAM physical layer error messages including at least one of OAM flow F3 messages and P-RDI and P-AIS messages indicating a detected loss of signal, loss of frame and/or loss of cell delineation.



5. ATM-Network comprising at least two interconnected ATM Network Elements (ATM NE-1) according to claim 1.



6. Method for automatic protection switching of a Transmission Convergence Sublayer SubNetwork Connection between a first ATM Network Element (ATM NE1) and a second ATM Network Element (ATM NE2) in an ATM network, each ATM Network Element (ATM NE1,NE2) comprising at least one working Transmission Convergence Sublayer entity (TCS-W) as part of a first physical line interface and at least one protection Transmission Convergence Sublayer entity (TCS-P) as part of a second physical line interface, said working Transmission Convergence Sublayer entities (TCS-W) of first and second ATM Network Element (ATM NE1,NE2) and said protection Transmission Convergence Sublayer entities (TCS-P) of first and second ATM Network Element (ATM NE1,NE2) enabling independent Transmission Convergence Sublayer SubNetwork Connections via different transmission paths, each of the Transmission Convergence Sublayer entities (TCS-W,TCS-P) being addressable within the first and the second ATM Network Element (ATM NE) respectively via a dedicated UTOPIA address, and each Transmission Convergence Sublayer entity (TCS-W,TCS-P) providing physical layer error messages indicating failures in the Transmission Convergence Sublayer SubNetwork Connection, the method comprising the steps of:



as long as no failure in the used Transmission Convergence Sublayer SubNetwork Connection is indicated by a provided physical layer error message,



(ii)

ISDN access available is the Primary Rate Interface (PRI), which is carried over an E1 (2048 kbit/s) in most parts of the world. An E1 is 30 'B' channels of 64 kbit/s, one 'D' channel of 64 kbit/s and a timing and alarm channel of 64 kbit/s. In North America PRI service is delivered on one or more T1 carriers (often referred to as 23B+D) of 1544 kbit/s (24 channels). A PRI has 23 'B' channels and 1 'D' channel for signalling (Japan uses a circuit called a J1, which is similar to a PRI). Inter-changeably but incorrectly, a PRI is referred to as T1 because it uses the T1 carrier format. A true T1 or commonly called 'Analog T1' to avoid confusion uses 24 channels of 64 Kbit/s of in band signaling. Each channel uses 56 kb for data and voice and 8 kb for signaling and messaging. PRI uses out of band signaling which provides the 23 B channels with clear 64 kb for voice and data and one 64 kb 'D' channel for signaling and messaging. In North America, Non-Facility Associated Signalling allows two or more PRIs to be controlled by a single D channel, and is sometimes called "23B+D + n*24B". D-channel backup allows for a second D channel in case the primary fails. NFAS is commonly used on a T3.



PRI-ISDN is popular throughout the world, especially for connection of PSTN circuits to PBXs.



PSTN is the acronym for Public Switched Telephone Network and is sometime refered to as POTS or Plain Old Telephone Service. PSTN actually refers to the telephone network i.e. PBX to PBX while POTS refers to the connection to a user as in residential or commercial. While the North American PSTN can use PRI or Analog T1 format from PBX to PBX the POTS or BRI can be delivered to a buisiness or residence. North American PSTN can actually connect from PBX to PBX via Analog T1, T3, PRI, OC3, etc...



Even though many network professionals use the term "ISDN" to refer to the lower-bandwidth BRI circuit, in North America by far the majority of ISDN services are in fact PRI circuits serving PBXs



A data communications network with at least one PoP maintains a local cache database associated with each AAA service at the PoP on the data communications network. Each local database contains a group identification such as a domain identification corresponding to a group of users or an FQDN specifying a group of one individual, a maximum number of B-Channels to provide the group of users at the PoP and a dynamic B-Channel session count corresponding to active B-Channel connections currently provided to the group of users at the PoP. Actions are taken when the group attempts to exceed the maximum number of B-Channels by more than a predetermined number. The actions may include assessing extra charges, denying access, and sending warning messages to appropriate recipients. The local database may be synchronized by publishing B-Channel connection and disconnection events to all subscribing local databases. For proxy authentication users, the authentication information is published to the local caches of each AAA service at the PoP upon the first log-in of the user so as to avoid the need to proxy each successive connection authentication to a remote AAA service.



(iii)



Packet switching is a digital networking communications method that groups all transmitted data – regardless of content, type, or structure – into suitably-sized blocks, called packets. Packet switching features delivery of variable-bit-rate data streams (sequences of packets) over a shared network. When traversing network adapters, switches, routers and other network nodes, packets are buffered and queued, resulting in variable delay and throughput depending on the traffic load in the network.



Packet switching contrasts with another principal networking paradigm, circuit switching, a method which sets up a limited number of dedicated connections of constant bit rate and constant delay between nodes for exclusive use during the communication session. In case of traffic fees, for example in cellular communication, circuit switching is characterized by a fee per time unit of connection time, even when no data is transferred, while packet switching is characterized by a fee per unit of information.



Cell Switching is similar to packet switching, except that the switching does not necessarily occur on packet boundaries. This is ideal for an integrated environment and is found within Cell-based networks, such as ATM. Cell-switching can handle both digital voice and data signals.



Cell Switching works similar than packet switching. The differences between both are the following:



* All information -data, voice, video- is transported from the origin-node to the end-node in small and constant-size packets (in traditional packet switching the packet size is variable) - 53 octets - called cells.

* Just lightened protocols are used in order to allow nodes fast switching. As a drawback protocols will be less efficient.

* Signaling is completely separated from information flow in contrast to packet switching in which both, information and signaling are mixed.

* Arbitrary binary rate traffic flows can be integrated in the same network.