Sub Code:EC2352 Sub Name: Computer Networks Dept: ECE Sem .
Sub Code:EC2352Sub Name: Computer NetworksDept: ECESem/Year: VI/IIIUNIT-IVPART-A1. What is three way hand shaking? (AUC MAY’13)2. Differentiate constant bit rate and variable bit rate. (AUC MAY’13)VBR dynamically adjusts compression ratio according to content, to achive best recordingquality.CBR uses a rather fixed compression ratio, set to use a given space.3. Explain the main idea of UDP. (AUC NOV’12)UDP is a very simple protocol using a nrinimum of overhead. Ifa process wants to send a small message and does not care much about reliability, it canuse UDP. Sending a small message by using UDP takes much less interaction betweenthe sender and receiver than using TCP or SCTP.4. What is meant by quality of service.(AUC NOV’12)A.JOYCE-AP/CSE –MAHALAKSHMI ENGINEERING COLLEGEPage 1
The quality of service defines a set of attributes related to the performance of the connection.For each connection, the user can request a particular attribute each service class is associatedwith a set of attributes.5. What are the duties of transport layer?( AUC NOV’11) The services provided by the transport layerEnd-to- end deliveryAddressing Reliable deliveryFlow controlMultiplexing6. What is meant by segment?( AUC NOV’11)A network segment is a portion of a computer network that is separated from the rest of thenetwork by a device such as a repeater, hub, bridge, switch or router. Each segment cancontain one or multiple computers or other hosts.The type of segmentation differs according to the type of device used. For example, a bridgeseparates collision domains, while a router separates both collision domains and broadcastdomains.7. The transport layer creates the connection between source and destination. What arethe three events involved in the connection?For security , the transport layer may create a connection between the two end ports. Aconnection is a single logical path between the source and destination that is associated with allpackets in a message. Creating a connection involves three steps:Connection establishmentData transfer & Connection release8. What is meant by congestion?Congestion in a network occur if user send data into the network at a rate greater than thatallowed by network resources9. Why the congestion occur in network?Congestion occur because the switches in a network have a limited buffer size to store arrivedpackets.10. How will the congestion be avoided?The congestion may be avoided by two bitsBECN - Backward Explicit Congestion NotificationFECN - Forward Explicit Congestion NotificationA.JOYCE-AP/CSE –MAHALAKSHMI ENGINEERING COLLEGEPage 2
11. What is the function of BECN BIT?The BECN bit warns the sender of congestion in network. The sender can respond to thiswarning by simply reducing the data rate.12. What is the function of FECN?The FECN bit is used to warn the receiver of congestion in the network. The senderand receiverare communicating with each other and are using some types of flow control at a higher level.13. What are the two categories of QoS attributes?The two main categories areUser OrientedNetwork Oriented14. List out the user related attributes?User related attributes are SCR – Sustainable Cell Rate PCR – Peak Cell RateMCR- Minimum Cell RateCVDT – Cell Variation Delay Tolerance15. What are the networks related attributes?The network related attributes are, Cell loss ratio (CLR)Cell transfer delay (CTD)Cell delay variation (CDV) Cell error ratio (CER)16. What are the four aspects related to the reliable delivery of data?The four aspects are, Error control Sequence controlLoss controlDuplication control17. What is function of transport layer?The protocol in the transport layer takes care in the delivery of data from one applicationprogram on one device to an application program on another device. They act as a link betweenthe upper layer protocols and the services provided by the lower layer.18. What are the duties of the transport layer?The services provided by the transport layer End-to- end delivery Addressing Reliable delivery Flow control Multiplexing19. What is frame?A frame consists of one complete cycle of time slots, including one or more slot dedicated toeach sending device.A.JOYCE-AP/CSE –MAHALAKSHMI ENGINEERING COLLEGEPage 3
PART-B1. Explain the following characteristics(AUC MAY’13)(i) Reliability(ii)Delay(iii) Jitter(iv) BandwidthReliabilityReliability is a characteristic that a flow needs. Lack of reliability means losing a packet oracknowledgment, which entails retransmission. However, the sensitivity ofapplication programsto reliability is not the same. For example, it is more important that electronic mail, file transfer,and Internet access have reliable transmissions than telephony or audio conferencing.DelaySource-to-destination delay is another flow characteristic. Again applications can tolerate delayin different degrees. In this case, telephony, audio conferencing, video conferencing,and remotelog-in need minimum delay, while delay in file transfer or e-mail is less important.JitterJitter is the variation in delay for packets belonging to the same flow. For example, if fourpackets depart at times 0, 1, 2, 3 and arrive at 20, 21, 22, 23, all have the same delay, 20 unitsof time. On the other hand, if the above four packets arrive at 21, 23, 21, and 28, they will havedifferent delays: 21,22, 19, and 24.For applications such as audio and video, the first case is completely acceptable; the secondcase is not. For these applications, it does not matter if the packets arrive with a short or longdelay as long as the delay is the same for all packets. For this application, the second case isnot acceptable.Jitter is defined as the variation in the packet delay. High jitter means the difference betweendelays is large; low jitter means the variation is small.BandwidthDifferent applications need different bandwidths. In video conferencing we need to send millionsof bits per second to refresh a color screen while the total number of bits in an e-mail may notreach even a million.2. (i) Explain how the connection is established and released in TCP with a neat sketch.A TCP connection begins with a client doing an active open to a server.Assuming that the server had earlier done a passive open, the two sides engage in anexchangeofmessagestoestablishtheconnection.Only afterthisconnectionestablishment phase is over do the two sides begin sending data. Likewise, as soon as aparticipant is done sending data, it closes one direction of the connection, which causesA.JOYCE-AP/CSE –MAHALAKSHMI ENGINEERING COLLEGEPage 4
TCP to initiate a round of connection termination messages.Connection setup is an asymmetric activity (one side does a passive open and theother side does an active open) connection teardown is symmetric (each side has to closethe connection independently). Therefore it is possible for one side to have done a close,meaning that it can no longer send data but for the other side to keep the other half of thebidirectional connection opens and to continue sending data.THREE WAY HANDSHAKES:The algorithm used by TCP to establish and terminate a connection is called athree way handshake. The client (the active participant) sends a segment to the quencenumberitplanstouse(flag SYN,SequenceNum x).The server then responds with a single segment that both acknowledges theclient‟ s sequence number (Flags ACK, Ack x 1) and states its own beginningsequence number (Flags SYN, SequenceNum y).That is, both the SYN and ACK bits are set in the Flags field of this second message.Finally, the client responds with a third segment that acknowledges the server‟ s sequenceA.JOYCE-AP/CSE –MAHALAKSHMI ENGINEERING COLLEGEPage 5
number Flags ACK, Ack y 1).(ii) Explain the default timer mechanism followed in TCP. (AUC MAY’13) TCP maintains four (4) timers for each connection:–Retransmission Timer: –Persist Timer –Ensures that window size information is transmitted even if no data istransmittedKeepalive Timer –The timer is started during a transmission. A timeout causes aretransmissionDetects crashes on the other end of the connection2MSL TimerMeasures the time that a connection has been in the TIME WAIT stateRetransmission Timer (RT Timer)Setting the RT timer When a segment is sent and RT timer is not running, start RT timer with RTO value Turn off RT timer, when all data is acknowledged When an ACK is received for new data, reset the RT timer to RTO valueRT timer expires Retransmit the earliest segment that has not been acknowledged Double value of RTO (see Karn’s rule) Start the RT timer with RTO valueTCP Persist Timer Assume the window size goes down to zero and the ACK that opens the window getslost If ACK (see figure) is lost, both sides are blocked. Persist Timer: Forces that the sender periodically queries the receiver about its windowsize (window probes)A.JOYCE-AP/CSE –MAHALAKSHMI ENGINEERING COLLEGEPage 6
ReceiverBuffer04K2K SeqNo 02KWin 2048AckNo 2048Sender blocked2K SeqNo 20484Kin 0WAckNo 40963KACK islostWin 1024AckNo 4096 The persist timer is started by the sender when the sliding window is zero Persist timer uses exponential backoff (initial value is 1.5 seconds) rounded to the range[5 sec, 60sec] So the time interval between timeouts are at: 5, 5, 6, 12, 24, 48, 60, 60, . The window probe packet contains one byte of data (TCP can do this even if the windowsize is zeroA.JOYCE-AP/CSE –MAHALAKSHMI ENGINEERING COLLEGEPage 7
Timeout(5 sec)Win 0AckNo 4096Probe1 byte SeqNo 4096Timeout(5 sec)Win 0AckNo 4096Probe1 byte SeqNo 4096Timeout(6 sec)Win 0AckNo 4096Probe1 byte SeqNo 4096Win 1024AckNo 40961 KBSeqNo 5120TCP Keepalive Timer When a TCP connection has been idle for a long time (1 min – 2 hours), a Keepalivetimer reminds a station to check if the other side is still there. A segment wihtout data is sent if the connection has been idle for 2 hours Assume a probe has been sent from A to B:(1) B is up and running:B responds with an ACK(2) B has crashed and is down:A will send 10 more probes, each 75seconds apart. If A does not get a response, it will close the connection(3) B has rebooted: B will send a RST segment(4) B is up, but unreachable: Looks to A the same as (2)3. Explain the congestion control technique in TCP.(AUC NOV’12)TCP CONGESTION CONTROLThe internet was suffering from congestion collapse-hosts would send their packetsinto the internet fast as the advertised window would allow, congestion would occur atsome router(causing packets to be dropped), & the hosts would time to out & retransmitstheir packets, resulting in even more congestion.A.JOYCE-AP/CSE –MAHALAKSHMI ENGINEERING COLLEGEPage 8
The idea of TCP congestion control is for each source to determine how muchcapacity is available in the network, so it knows how many packets it can safely have intransit. Once a given source has this many packets in their transit, it uses the arrival of anACK as a signal that one of its packets has the left thenetwork, & that it is therefore safe toinsert a new packet into the network without adding to the level of congestion. Byusing ACKs to pace the transmission of packets, TCP is said to self-clocking.ADDITIVE INCREASE/MULTIPLICATIVE DECREASE:TCP maintains a new state variable for each connection, called CongestionWindow, which is used by the source to limit how much data it is allowed to have in transitat a given time .The congestion window is congestion control‟ s counterpart to flow control‟ sadvertised window. TCP is modified such that the maximum number of bytes ofunacknowledgeddata allowed is now theminimum of the congestion window and the advertised window .TCP‟ s effective window isrevised as follows:Max Window MIN (Congestion Window, Advertised Window) Effective Window Max Window – (Last Byte Sent- Last Byte Acked)The problem, of course, is TCP comes to learn an appropriate value for CongestionWindow . Unlike the Advertised Window, which is sent by the receiving side of theconnection, there is no one to send a suitable Congestion Window based on the level ofcongestion it perceives to exist in the network. This involves decreasing the congestiongoes up & increasing the congestion window when the level of congestion goes down.Taken together, the mechanism is commonly called additive e additive increase mechanism just described is the right approach to use when thesource is operating close to the available capacity of the network, but it takes too long toramp up a connection when it is starting from scratch. TCP therefore provides the secondmechanism, ironically called slow chat that is used to increase the congestion window rapidlyfrom the cold start. Slow start effectively increases the congestion window exponentially, ratherthan linearly.The source starts out by setting Congestion Window to one packet. When the ACKA.JOYCE-AP/CSE –MAHALAKSHMI ENGINEERING COLLEGEPage 9
for this packet arrives, TCP adds 1 to Congestion Window and then sends two packets.Upon their receiving the corresponding two ACKs, TCP increments the CongestionWindow by 2-one for each ACK –and next sends four packets. The end result is that TCPeffectively doubles the number of packets it has in transit every RTT.There are actually two different situations in which slow start runs. The first is at thevery beginning of a connection, at which time the source has no idea how many packets it isgoing to be able to have in transit at and given time. In this situation, slow start continues todouble Congestion Window each RTT until there is and loss, at which time a timeout cause‟ smultiplicative decrease to divide Congestion Window by 2.The second situation in which slow start is a bit more subtle; it occurs whenthe connection goes dead while waiting for a timeout to occur. Recall how TCP „s slidingwindow algorithm works – when the packet is lost, the source eventually reaches a pointwhere it is sent as much data as the advertised window allows, and so it blocks whilewaiting for an ACK that will not arrive. Eventually, a timeout happens, but by this time thereare no packets in transit; meaning that the source will receive no ACKs to “clock” thetransmission of new packets. The source will instead receive a single commutativeACK that reopens the entire advertised window, but as explained above, the source thenuses slow start to restart the flow of data rather than dumping a whole window‟ s worth ofdata on the network all at once.FAST RETRANSMIT AND FAST RECOVERY:The idea of fast retransmit is straight forward. Every time a data packet arrives at thereceiving side, the receiver responds with an acknowledgement, even if this sequencenumber has already been acknowledged. Thus, when a packet arrives out of order – that is,TCP cannot yet acknowledge the data the packet contains because earlier data has not yetarrived – TCP resends the same acknowledgement it sent the last time. This secondtransmission of the acknowledgement is called a duplicate ACK .When the sending sidesees a duplicate ACK , it knows that the other side must have received a packet out of order,which suggests that an earlier packet has might have been lost. Since it is also possible thatthe earlier packet has only been delayed rather than lost , the sender waits until it sees someA.JOYCE-AP/CSE –MAHALAKSHMI ENGINEERING COLLEGEPage 10
number of duplicate ACKs and then retransmits the missing packet. In practice, TCP waitsuntil it has seen three duplicate ACKs before retransmitting the packet.4. Explain the segment formats for TCP and UDP.(AUC NOV’12)The segment consists of a 20- to 60-byte header, followed by data from the applicationprogram.Source port address. This is a 16-bit field that defines the port number of the applicationprogram in the host that is sending the segment. This serves the same purpose as the sourceport address in the UDP header.Destination port address. This is a 16-bit field that defines the port number of the applicationprogram in the host that is receiving the segment. This serves the same purpose as thedestination port address in the UDP header.Sequence number. This 32-bit field defines the number assigned to the first byte of datacontained in this segment.Acknowledgment number. This 32-bit field defines the byte number that the receiver of thesegment is expecting to receive from the other party.Header length. This 4-bit field indicates the number of 4-byte words in the TCP header. Thelength of the header can be between 20 and 60 bytes. Therefore, the value of this field can bebetween 5 (5 x 4 20) and 15 (15 x 4 60).Reserved. This is a 6-bit field reserved for future use.Control. This field defines 6 different control bits or flags as shown in Figure or more of thesebits can be set at a time.Control fieldA.JOYCE-AP/CSE –MAHALAKSHMI ENGINEERING COLLEGEPage 11
Window size. This field defines the size of the window, in bytes, that the other party mustmaintain. Note that the length of this field is 16 bits, which means that the maximum size of thewindow is 65,535 bytes.Checksum. This 16-bit field contains the checksum. The calculation of the checksum for TCPfollows the same procedure as the one described for UDPUrgent pointer. This l6-bit field, which is valid only if the urgent flag is set, is used when thesegment contains urgent data. It defines the number that must be added to the sequencenumber to obtain the number of the last urgent byte in the data section of the segment.Options. There can be up to 40 bytes of optional information in the TCP headerUDP:A.JOYCE-AP/CSE –MAHALAKSHMI ENGINEERING COLLEGEPage 12
The header for an end-to-end protocol that implements this demultiplexingfunction typically contains an identifier (port) for both the sender (source)and thereceiver(destination)of the message. The UDP port field is only 16bits long. This means thatthere are up to 64K possible ports, clearly not enough to identify all the processes on all thehosts in the Internet. Fortunately, ports are not interpreted across the entire Internet,but only on a single host. That is, a process is really identified by a port on some particularhost- port, host pair. In fact, this pair constitutes the demultiplexing key for the UDPprotocol.5. Discuss the congestion control mechanism and transmission control protocol withneat sketches ( AUC NOV’11)Congestion control refers to techniques and mechanisms that can either preventcongestion,before it happens, or remove congestionOpen-Loop Congestion ControlIn open-loop congestion control, policies are applied to prevent congestion before it happens. Inthese mechanisms, congestion control is handled by either the source or thedestination.Retransmission PolicyRetransmission is sometimes unavoidable. If the sender feels that a sent packet is lost orcorrupted, the packet needs to be retransmitted. Retransmission in general may increasecongestion in the network. However, a good retransmission policy can prevent congestion. Theretransmission policy and the retransmission timers must be designed to optimize efficiency andat the same time prevent congestionWindow PolicyThe type of window at the sender may also affect congestion. The Selective Repeat window isbetter than the Go-Back-N window for congestion control. In the Go-Back-N window, when thetimer for a packet times out, several packets may be resent, although some may have arrivedsafe and sound at the receiver. This duplication may make the congestion worse. The SelectiveRepeat window, on the other hand, tries to send the specific packets that have been lost orcorrupted.Acknowledgment PolicyA.JOYCE-AP/CSE –MAHALAKSHMI ENGINEERING COLLEGEPage 13
The acknowledgment policy imposed by the receiver may also affect congestion. If the receiverdoes not acknowledge every packet it receives, it may slow down the sender and help preventcongestion. Several approaches are used in this case. A receiver may send an
Sub Code:EC2352 Sub Name: Computer Networks Dept: ECE Sem/Year: VI/III UNIT-IV PART-A 1. What is three way hand shaking? (AUC MAY’13) 2. Differentiate constant bit rate and variable bit rate. (AUC MAY’13) VBR dynamically adjusts compression ratio according to content, to achive best recording .
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