Computer Network UNIT 3
Computer Network UNIT 3
UNIT - 3
Point-to-point Connections
The point-to-point scheme provides separate communication channels for each pair of
computers. When more than two computers need to communicate with one another, the
number of connections grows very quickly as number of computer increases. Above figure
illustrates that two computers need only one connection, three computers need three
connections and four computers need six connections.
By: Nihal Kumar
The point-to-point scheme provides separate communication channels for each pair of
computers. When more than two computers need to communicate with one another, the
number of connections grows very quickly as number of computer increases. Above figure
illustrates that two computers need only one connection, three computers need three
connections and four computers need six connections.
As the Figure illustrates that the total number of connection grows more rapidly than the
total number of computers. Mathematically, the number of connection needed for N
computers is proportional to the square of N.
Adding the Nth computer requires N·l new connections which becomes a very expensive
option. Moreover, many connections may follow the same physical path. Figure shows a
point- to-point connection for five computers located at two different locations, say, ground
and first floor of a building.
As there are five PCs, total ten connections will be required for point-to-point connection. Out
of these ten connections six are passing through the same locution and thereby making
point-to-point connection an expensive one. Increasing the PC by one in the above
configuration at location 2 as shown in Figure will increase the total number of connections to
fifteen. Out ‘of these connections eight connections will pass through the same area.
By: Nihal Kumar
Routing algorithms
The main function of the network layer is routing packets from the source machine to the
destination machine. Routing algorithm can be grouped into two major classes. Nonadaptive and
Adaptive algorithms.
3) When the network is booted the 3) The routers are not downloaded.
routers are downloaded.
4) This is a static routing. 4) This is a dynamic routing.
B 7 B(2,A) C(∞,-)
2 3
E 2 F
2 3C E(∞,-)
A D A F(∞,-) D(∞,-)
1 2
6 4
2
G H G(6,A) H(∞,-)
(a) (b)
E(4,B) E(4,B)
A F(∞,-) D(∞,-) A F(6,E) D(∞,1)
E(4,B) E(4,B)
A F(6,E) D(∞,-) A F(6,E) D(∞,-)
One-way of measuring path length is the number of hops. Using this metric, the paths
ABC and ABE are equally long. (Two hops).
Another metric is the Geographic distance in Kilometers. ABC is clearly longer than ABE.
Many other metrics are also possible besides hops and physical distance. Each are could
be labeled with the mean queuing and transmission delay for some standard test packets as
determined by hourly test runs. With this graph labeling, the shortest path is the fastest path,
rather than the path with the fewest arc or kilometers.
In most 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.
The shortest path can be calculated using Dijkstra method. Each node is labeled with its
distance from the source 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. Initially all labels are tentative. When it is discovered that a label
represents the shortest path from the source to that node, it is made permanent and never
changed thereafter.
In the above diagram, let the weights represents the distance. To find out the shortest path
from A to D. We start by marking A as permanent. The examine each one with the distance to A,
relabeling each one with the distance to A. Whenever a node is relabeling also label it with the
node from which the probe was made. After examing each of the nodes adjacent to A, examine
all the tentatively labeled nodes in the whole graph and make the one with the smallest label
permanent. This one becomes the new working node.
The same procedure is adopted to all the nodes and the shortest path is found.
Flooding:
This is a static algorithm. In this, every incoming packet is sent out on every outgoing line except
the one it arrived on. Flooding will generate vast numbers of duplicate packets, some measures
have to take to dump the duplicate packets. One such measure is to have a hop counter
contained in the header of each packet, which is decremented at each hop, with the packet being
discarded when the counter reaches zero. The hop counter should be initialized to the length of
the path from source to destination. If the sender does not know how long the path is it can
initialize the counter to full diameter of the subnet.
A variation of flooding is ‘Selective Flooding’. In this the routers do not send every
incoming packet on every line, instead only on those lines that are going approximately in the
right direction which leads to the destination.
By: Nihal Kumar
Advantages
1) In military applications, where large numbers of routers are blown, flooding is desirable.
2) In Distributed database applications, it is some times necessary to update all the databases
concurrently, in which flooding is useful.
3) It is used as a metric against which other routing algorithms are compared.
4) Flooding chooses the shortest path, because it chooses all possible path in parallel.
Flow-based Routing:
The flooding and shortest path algorithm takes the topology in to account. Flow based
routing algorithm uses both topology and load for routing.
In some networks the mean data flow between each pair of nodes is relatively stable and
predictable. The average traffic is known in advance and to a reasonable approximation, constant
in time, it is possible to analyze the flows mathematically to optimize routing.
The basic idea behind this, for a given line, if the capacity and average flow are known, it
is possible to compute the mean packet delay on that line. From the mean delays on all the
lines, the mean packet delay for the whole subnet.
To use this technique, certain information must be known in advance. 1) About the subnet
topology 2) about the traffic 3) the line capacity 4) a routing algorithm
The fig.(b) gives the information packets /sec go from source i to destination j. Given this
information, it is easy to calculate the total in line for i i.e. i . Using the traffic from the source to
destination the mean number of packets/sec on each line. ci can be calculated, assuming a
mean packet size 1/ . The mean delay for each line can be derived where 1/ is the mean
packet size in bits, is the mean flow in packets/sec.
With a capacity c = 25 packets/sec and an actual flow = 14 packets/sec, the mean delay is 91
m sec. When =0, the mean delay is 40m sec. With this example we can say the delay depends
on both queuing and service time.
The mean delay time for the entire subnet can be calculated as the sum of each of the eight lines,
with the weight being the fraction of the total traffic using that line.
By: Nihal Kumar
Destination
A B C D E F
9 4 1 7 4
A AB ABC ABFD AE AEF
9 8 3 2 4
B 20 C AB BC BFD BFE BF
B
20 10 4 8 3 3 2
Source
C CBA CB CD CE CEF
A D
20 20 1 3 3 3 4
D DFBA DFB DC DCE DF
20 10
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F E EA EFB EC ECD EF
50 E
(a) 4 4 4 4 4
F FEA FB FEC FD FE
(b)
Distance Vector Routing:
This is a dynamic routing algorithm. This algorithm operates by having each router
maintain a table (i.e. a vector) giving the best known distance to each destination and which line
to use. These tables are updated by exchanging information with the neighbors.
The routing table indexed by and containing one entry for each router in the subnet. This
entry contains two parts: The preferred outgoing line to use for the destination and an estimate of
time or distance to that destination. The metric used might be number of hops, time delay in
msec, total number of packets queued along the path or something similar.
The router is assumed to know the distance to each of its neighbors. If the metric is hops,
the distance is just one hop. If the metric is queue length, the router examines each queue. If the
metric is delay the router can measure it directly with a special ECHO packets.
Consider an example, in which the delay is used as metric and the router knows the delay
to each of its neighbors. Once every T msec each router send to each neighbor a list of its
estimated delays to each destination. It also receives a similar list from each neighbor. Let xi
being x’s estimate of how long it takes to get router ‘i’. If the router knows that the delay to x is ‘m’
m sec. To get router i via x is (xi +m) m sec. By performing this calculation for each neighbor, a
router can find out which estimate is the best and use that estimate and the corresponding line in
its new routing table.
By: Nihal Kumar
New estimated
delay from J
Router
A I H K Line
A B
C D
A 0 24 20 21 8 A
B 12 36 31 28 20 A
G 25 18 19 36 28 I
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F D 40 27 8 24 20 H
E 14 7 30 22 17 I
F 23 20 19 40 30 I
I J K L
G 18 31 6 31 18 H
H 17 20 0 19 12 H
Subnet
I 21 0 14 22 10 I
J 9 11 7 10 0 -
K 24 22 22 0 6 K
L 29 33 9 9 15 K
JA JI JH JK
delay delay delay delay New
is is is is routing
8 10 12 6 table for J
Using hierarchy the path lengths will be increased. For Ex. The best path from 1A to 5C is via
region 2. But using hierarchy routing all traffic to region 5 goes via region 3, because it is the best
for most destination in region 5.
Ex. Consider a 720 routers subnet.
Without hierarchy each router required 720 entries. Total entries will be
720 x 720, with hierarchy, if the subnet is portioned into 24 regions and
30 routers/region, then each router needs 30+23 = 53 entries only.
If a 3-level hierarchy used, with 8 clusters, each contains 9 regions and
10 routers/region. Each router needs 10 + 8 + 7 = 25 entries
For a ‘N’ router subnet the optimal number of routers = lnN
The total no. of entries /router = elnN
Broadcast Routing:
In some applications, hosts need to send messages to many or all other hosts. For Ex.
Weather reports, stock market, updates etc. Sending a packet to all destinations simultaneously is
called ‘Broad Casting’.
1) One method is sending distinct packet to each destination by the source. This method wastes
the bandwidth and also requires the source to have a complete list of all destinations.
2) The second is using Flooding technique. This generates too many packets and consumes too
much bandwidth.
3) Another method is multi destination routing. In this each packet contains either a list of
destinations or a bit map indicating the desired destinations. When a packet arrives at a router,
the router checks all the destinations to determine the set of output lines that will be needed. The
router generates a new copy of the copy for each output line to be used and includes in each
packet only those destinations that are to use the line. This routing is like separately addressed
packets except that several packets must follow the same route.
4) Another algorithm which uses the spanning tree. A spanning tree is a subnet of the subnet that
includes all the routers but contains no loop. If each router knows which of its lines belong to the
spanning tree, it can copy an incoming broadcast packet onto all spanning tree lines except the
one it arrived on. This method makes use of bandwidth excellently and generates minimum no. of
packets necessary to do the job. The only disadvantage is that each router must have knowledge
of some spanning tree.
5) One more algorithm is an attempt to approximate the behavior of the previous one, even when
the routers do not know anything at all about spanning trees. The idea is remarkably simple once
it has been pointed out. When a broadcast packet arrives at a router, the router checks to see if
the packet arrived on that line that is normally used for sending packets to the source of the
broadcast. If so, there is an excellent chance that the broadcast packet itself followed the best
By: Nihal Kumar
route from the router and is therefore the first copy to arrive at the router. This being the case, the
router forwards copies of it onto all lines except the one it arrived on. If, however, the broadcast
packet arrived on a line other than the preferred one for reaching the source the packet is
discarded as a likely duplicate.
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By: Nihal Kumar
Multicast Routing :
For some applications, it is necessary for one process to send a message to all other members of
the group. If the group is small, it can just send each other member a point-to-point message. If
the group is large this strategy is expensive. Some times broad casting is used, but using broad
casting is used, but using broadcasting to inform 1000 machines on a million node network is
inefficient because most receivers are not interested in the message. Thus it is needed to send
message to well-defined groups. Sending message to such a group is called ‘multicasting’.
To do multicasting, group management is required. Some way is needed to create and
destroy groups and for processes to join and leave groups. When process joins a group, it informs
its host of this fact. It is important that routers know which of their hosts belong to which group.
Either hosts most inform their routers about change in group membership or routers must query
their hosts periodically. Routers tell their neighbors, so the information propagates through the
subnet.
To do multicast routing, each router computes a spanning tree covering all other routers in the
subnet. When a process sends a multicast packet, to a group, the first router examines its
spanning tree and prunes it, removing all lines that do not lead to hosts that are members in the
group. Multicast packets are forwarded. Only along the appropriate spanning tree.
1. Three or four input lines and only one output line queue will build up.
If there is insufficient memory to hold all of them, packets will lost.
Adding infinite memory congestion gets worse, because by the time packets get to the
front of the queue, the time out and duplicates have been sent.
2. Slow processors (routers) can cause congestion.
A slow processor perform the book keeping tasks very slow, queues will build up.
By: Nihal Kumar
Ex:
Consider a network with a capacity of 1000Gbps on which a super computer is trying to
transfer a file to a personal computer at 1Gbps.Here a flow control is needed.
Consider a network with 1Mbps lines and 1000 large computers, more than half are trying
to transfer files a 100kbps to the other half. The problem is here is the total offered traffic
exceeds than the network handle.
Imagine a bucket with a small hole in the bottom. No matter at what rate water enters the bucket,
the outflow is at a constant rate, , when there is any water in the bucket, and zero when the
bucket is empty. Also, once the bucket is full, any additional water entering it spills over the sides
and is lost.
Host
Computer
Packet
Unregulated
Flow
Regulated
flow
Network
The same idea can be applied to packets, as shown in fig. Conceptually, each host is connected
By: Nihal Kumar
to the network by an interface containing a leaky bucket, that is, a finite internal queue. If a packet
arrives at the queue when it is full, the packet is discarded. In other words, if one or more
processes within the host try to send a packet when the maximum numbers are already queued,
the new packet is unceremoniously discarded. This arrangement can be built into the hardware
interface or simulated by the host operating system.
The host is allowed to put one packet per clock tick onto the network. Again, this can be enforced
by the interface card or by the operating system. This mechanism turns an uneven flow of packets
from the user processes inside the host into an even flow of packets onto the network, smoothing
out bursts and greatly reducing the chances of congestion.
Implementing the original leaky bucket algorithm is easy. The leaky bucket consists of a finite
queue. When a packet arrives, if there is room on the queue it is appended to the queue;
otherwise, it is discarded. At every clock tick, one packet is transmitted (unless the queue is
empty).
Host Host
Computer Computer
One token
is added
to the bucket The bucket
holds
every T
tokens
In this algorithm, the leaky bucket holds tokens, generated by a clock at the rate of one token ever
T sec. In figure (a), we see a bucket holding three tokens, with five packets waiting to be
transmitted. For a packet to be transmitted, it must capture and destroy one token. In figure(b), we
see that three of the five packets have gotten through, but the other two are stuck waiting for two
more tokens to be generated.
The leaky bucket algorithm does not allow idle hosts to save up permission to send large bursts
later. The token bucket algorithm does allow saving, up to the maximum size of the bucket, n.
Another difference between the two algorithms is that the token bucket algorithm throws away
tokens when the bucket fills up but never discards packets. In contrast, the leaky bucket algorithm
discards packets when the bucket fills up.
Each router can easily monitor the utilization of its output lines and other resources. It can
estimate each line about the recent utilization of that line (u). Periodically a sample at the
instantaneous line utilization (f) can be mad and u updated.
unew = a uold + (1-a)f
Where a is constant determines how fast the router forgets recent history.
Whenever u moves above the threshold, the output line enters a ‘warning’ state. Each new
arriving packet is checked if its output line is warning state. If it is some action is taken.
Choke packets:
In this algorithm, the router sends a choke packet back to the source host. The original packet is
tagged so that it will not generate any more choke packets farther along the path and is then
forwarded in the usual way.
When the source host gets the choke packet, it is required to reduce the traffic sent to the
specified destination by X percent. Since other packets aimed at the same destination are
probably already under way and will generate yet more choke packets, the host should ignore
choke packets referring to that destination for a fixed time interval. After that period has expired,
the host listens for more choke packets for another interval. If one arrives, the line is still
congested, so the host reduces the flow still more and begins ignoring choke packets again. If no
choke packets arrive during the listening period, the host may increase the flow again.
The first choke packet causes the data rate to be reduced to 0.50 of its previous rate, the next
one causes a reduction to 0.25, and so on. Increases are done in smaller increments to prevent
congestion from reoccurring quickly.
By: Nihal Kumar
B C B C
A D A D
Heavy
flow
E F E F
Ch Ch
Ch Ch
Reduced
flow
(b)
Flow is still at
maximum rate
Ch-choke
Flow is reduced
(a)
By: Nihal Kumar
Load Shedding:
Load Shedding is a fancy way of saying that when routers are being inundated by packets that
they cannot handle, they just throw them away.
A router drowning in packets can just pick packets at random to drop, but usually it can do better
than that. Which packet to discard may depend on the applications running. For file transfer, an
old packet is worth more than a new one because dropping packet 6 and keeping packets 7
through 10 will cause a gap at the receiver that may force packets 6 through 10 to be
retransmitted (if the receiver routinely discards out-of –order packets). In a 12-packet file,
dropping 6 may require 7 through 12 to be retransmitted, whereas dropping 10 may require only
10 through 12 to be retransmitted. In contrast, for multimedia, a new packet is more important
than an old one. The former policy (old is better than new) is often called wine and the latter (new
is better than old) is often called milk.
A step above this in intelligence requires cooperation from the senders. For many applications,
some packets are more important than others. For example, certain algorithms for compressing
video periodically transmit an entire frame and then send subsequent frames as differences from
the last full frame. In this case, dropping a packet that is part of a difference is preferable to
dropping one that is part of a full frame. As another example, consider transmitting a document
containing ASCII text and pictures. Losing a line of pixels in some image is far less damaging
than losing a line of readable text.
Internetworking
When two or more networks are connected it is called Internet. There will be a variety of different
networks will always be around, for the following reasons.
1) Different networks will use different technologies like personal computers run TCP/IP,
mainframes run on IBM’s SNA.
2) As computers and networks get cheaper, the place where decisions get made moves
downwards in organizations.
3) As new hardware developments occur, new software will be created to fit the new hardware.
The purpose of interconnecting all these networks is to allow users on any of them to
communicate with users don all the other ones to allow users on any of them to access data on
any of them.
Networks differ in many ways. In the network layer the following differences can occur (fig.5.43).
By: Nihal Kumar
multi protocol router to the other. Only the multiprotocol router has to understand IP and WAN
packets.
The IP Protocol
At the network layer, the Internet can be viewed as a collection of subnet-works or Autonomous
systems that are connected together. The network layer protocol that used for Internet is Internet
Protocol (IP). Its job is to provide a best-efforts way to transport datagrams from source to
destination, without regard to whether or not these machines are on the same network or not
these are other networks in between them.
Communication in the Internet works as follows. Each datagram is transmitted, after getting from
Transport layer, through the Internet, possibly being fragmented into smaller units as it goes.
When all pieces finally get to the destination machine, they are reassembled by the network layer
into the original datagram.
Internetwork protocol
At the network layer, TCP/IP supports the internetwork protocol .IP, in turn, contains four
supporting protocols:ARP ,RARP ,ICMP,and IGMP.
IP is the transmission mechanism used by the TCP/IP protocols. It is an un –reliable and
connectionless datagram protocol – a best effort delivery service. This is like a post office service.
The post office does its best to deliver the mail but does not always succeed. If an unregistered
letter is lost. it is up to the sender or would recipient to discover the loss and rectify the problem.
The post office itself does not keep track of every letter and cannot notify a sender of loss or
damage. An example of a situation similar to pairing IP with a protocol that contains reliability
functions is a self addressed ,stamped postcard included in a letter mailed through the post office.
when the letter is delivered , the receiver mails the postcard back to the sender to indicate
success. If the sender never receives the postcard, he or she assumes the letter was lost and
sends out another copy.
Packets in IP layer are called Datagrams. A Datagram is a variable length packet(upto 65,536
bytes) consisting of two parts : Header and Data. The header can be from 20 to 60 bytes and
contains information essential to routing and delivery.
By: Nihal Kumar
Version The first field defines the version number of the IP. The current version is 4(IPv4),with
binary value 0100.
Header length (HLEN) The HLEN field defines the length of the header in multiples of four
bytes .The four bits can represent a number between 0 to 15,which,when multiplied by 4,gives a
maximum of 60 bytes.
Service Type. The service type field defines how datagram should be handled. It includes bits
that define the priority of the datagram. It also contains bits that specify the type of service the
sender desires such as the level of throughput, reliability, and delay.
Total Length The total length field defines the total length of the IP datagram. It is a two-byte
field (16 bits) and can define up to 65,535 bytes.
Identification The identification field is used in fragmentation. A datagram, when passing through
different networks, may be divided into fragments to match the network frame size. When this
happens, each fragment is identified with a sequence number in this field.
Flags The bits in the flags field deal with fragmentation (the datagram can or can not be
fragmented; can be first, middle, or last fragment; etc.).
Fragmentation offset The fragmentation offset is a pointer that shows the offset of the data in
the original datagram (if it is fragmented).
Time to live The time to live field defines the number of hops a datagram can travel before it is
discarded. The source host, when it creates the datagram, sets this field to an initial value. Then,
as the datagram travels through the Internet, router by router, each router decrements this value
by 1. If this value becomes 0 before the datagram reaches its final destination, the datagram is
discarded. This prevents a datagram from going back and forth forever between routers.
Protocol The protocol field defines which upper-layer protocol data are encapsulated in datagram
(TCP, UDP, ICMP etc.).
Header Checksum This is a 16-bit field used to check the integrity of the header, not the rest of
the packet.
Source address The source address field is a four-byte (32-bit) Internet address. It identifies the
original source of the datagram.
Destination address The destination address field is a four-byte (32-bit) Internet address. It
identifies the final destination of the datagram.
Options The options field gives more functionality to IP datagram. It can carry fields that control
routing, timing, management, and alignment.
ADDRESSING
In addition to the physical address the internet requires an additional addressing convention : an
address that identifies the connection of a host to its network.
By: Nihal Kumar
Each Internet address consists of 4 bytes defining three fields : class type,netid,and hosted.
These parts are varying lengths depending on the class of the address.
CLASSES
There are currently five different classes:
Class A :
This can accommodate more hosts since 3 bytes are reserved for HOSTID. Class A will begin
with 0 .
Class B :
This will start with 10 and Host id will have 2 bytes length.
Class C :
This will start with 110 and Hostid will have 1 byte length.
Class D:
This will start with 1110 . This is reserved for Multicast addresses.
Class E :
This is reserved for feature use and will start with 1111 .
CLASS A :
000
0 Netid Hostid
CLASS B:
10 Netid 000
Hostid
Class C :
Class D :
Class E :
Dotted-Decimal Notation:
To make 32 bit form shorter and easier to read, Internet addresses are usually written in decimal
form with decimal points separating the bytes – dotted – decimal notation.
128.11.3.31
From To
0.0.0.0 127.255.255.2555 Class A
From To
128.0.0.0
12 191.255.255.255 Class B
From To
From To
224.0.0.0 239.255.255.255 Class D
From To
240.0.0.0 255.255.255.2555 Class E
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Example:
a. 4.23.145.90
b. 227.34.78.7
c. 246.7.3.8
d. 129.6.8.4
e. 198.76.9.23
Example:
a. 4.23.145.90
b. 227.34.78.7
c. 246.7.3.8
d. 129.6.8.4
e. 198.76.9.23
TCP/IP supports four other protocols in the network layer :ARP,RARP,ICMP,and IGMP.
By: Nihal Kumar
The address resolution Protocol associates an ip address with physical address. On a typical
physical network, such as a LAN, each device on a link is identified by a physical or station
address usually imprinted on the network interface card.(NIC)
Physical address have local jurisdiction and can be changed easily. For example, if the NIC on
a particular machine fails, the physical address changes. The IP address, on the other hand ,have
universal jurisdiction and cannot be changed. ARP is used to find the physical address of the
node when its Internet address is known.
Anytime a host or a router needs to find the physical address of another host on its
network, it formats an ARP query packet that includes the IP address and broadcast it over the
network. Every host on the network receives and processes the ARP packet, but only the
intended recipient recognizes its internet address and sends back its physical address. The host
both to its cache memory and to the datagram header, then sends the datagram on its way.
Reverse Address resolution protocol(RARP)
The RARP allows a host to discover its internet address when it knows only its physical
address. The question here is ,why do we need RARP? A host is supposed to have its internet
address stored on its hard disk !
RARP works much like ARP. The host wishing to retrieve its internet address broadcasts an
RARP query packet that contains its physical address to every host on its physical network. A
server on the network recognizes the RARP packet and returns the host’s internet address.
Internet Control Message Protocol (ICMP)
The Internet control message protocol is a mechanism used by hosts and routers to send
notification of datagram problems back to the sender.
IP is an unreliable and connectionless protocol. ICMP allows IP to inform a sender if a
datagram is undeliverable. A datagram travels from router to router until it reaches one that can
deliver it to its final destination. If a router is unable to route or deliver the datagram because of
unusual conditions or due to congestion, ICMP allows it to inform the original source.
ICMP uses echo test/reply to test whether a destination is reachable and responding. It
also handles both control and error message, but its sole function is ti\o report problems, not
correction them. A datagram carries only source and destination address. For this reason ICMP
can send message only to the source, not to an intermediate router.
Ip addressing supports multicasting. All 32-bit IP addresses that start with 1110(class D) are
multicast addresses. With 28 bits remaining for the group address, more than 250 million
addresses are available for assignment. Some of these addresses are permanently assigned.
The IGMP has been designed to help a multicast router identify the hosts in a lan that are
members of a multicast group. It is a companion to the IP protocol.
Expected questions
1.What is the difference between the adaptive and non-adaptive routing algorithms.
2.Explain the shortest path routing algorithm.
3.Explain the services that are provided by the network layer.
4.Explain Flooding routing algorithm.
5.Explain the Distance Vector Routing algorithm.
6.What is the count – to – infinity problem?
7.Explain link state routing algorithm.
8.Explain the Hierarchical Routing algorithm.
9.Explain Broadcast Routing and Multicast Routing.
10.What is congestion? Give the general principles of congestion control?
11.Explain Open loop and Close loop solutions for congestion.
12.How traffic shaping will be done to control congestion?
13.Explain The Leaky Bucket algorithm.
14. Explain the Token Bucket algorithm.
15.How the congestion can be controlled in Virtual Circuits?
16.What is a Choke packet? Explain when a choke packet is used.
17.Expalin the IP protocol.
18What is meant by Load shedding and Jitter control?
19. Explain the ICMP and ARP.
20.Explain the different IP address formats. For a hierarchical routing with 4800 routers, what
region and cluster sizes should be chosen to minimize the size of routing table for a three-layer
hierarchy?
* * * * *
By: Nihal Kumar
Multiple choice Questions
1.Which OSI layer corresponds to the TCP-UDP layer?
a. physical b. data link c.network d. transport
2.Which OSI layer corresponds to the IP layer?
a. physical b. data link c.network d. transport
3. Which OSI layer(S) corresponds to the TCP/IP application layer?
a. application b. session c. presentation d.all of the above
4.Which IP address class has few hosts per network?
a. A b.B c.C d. D
5.For what does the data link layer look for as it sends a frame from one link to another?
a. hosted b.Ip address c. domain name d.station address
6.The purpose of ARP on a network is to find the ------- given the ---------
a. Intenet address, domain name
b. Intenet address , station address
c. Intenet address ,netid
d. station address ,Intenet address
7.Which of the following apply to UDP?
a. is unreliable and connectionless
b. contains destination and source port address
c. reports certain errors
15.when a host knows its physical address but not its IP address,it can use-----
a.ICMP b.IGMP c. ARP d. RARP
16.This transport layer protocol is connectionless.
a.UDP b.TCP c. FTP d.NVT
17. This transport layer protocol requires acknowledgement.
a. UDP b.TCP c. FTP d.NVT
18.Which of the following is default mask for the address 98.0.46.201?
a.255.0.0.0
b.255.255.0.0
c.255.255.255.0
d.255.255.255.255
19. Which of the following is default mask for the address 98.0.46.201
a.255.0.0.0
b.255.255.0.0
c.255.255.255.0
d.255.255.255.255
20. Which of the following is default mask for the address 190.0.46.201?
a.255.0.0.0
b.255.255.0.0
c.255.255.255.0
d.255.255.255.255
21.Change the following IP address from dotted-decimal notation to binary notation
a. 114.34.2.8 b.129.14.6.8 c.208.34.54.12 d.238.34.2.1 e. 241.34.2.8
22.Find the class of the following IP address
a.208.34.54.12 b.238.34.2.1 c. 129.14.6.8 d.241.34.2.8
23.Find the netid and hosted for the following IP address.
a.114.34.2.8 b.19.34.21.5 c. 23.67.12.1 d. 126.23.4.0
24. Find the netid and hosted for the following IP address.
a.129.14.6.8 b.132.56.8.6 c.23.67.12.1 d.190.12.67.9
25. Find the netid and hosted for the following IP address.
a. 192.8.56.2 b. 220.34.8.9 c.208.34.54.12 d. 205.23.67.8
26.Find the network address of the following IP addresses.
a. 114.34.2.8 b. 171.34.14.8 c. 192.8.56.2 d.226.7.34.5
f.226.7.34.5 f. 225.23.6.7 g. 245.34.21.5
27.Write the following masks in binary notation:
a.255 .255.255.0 b.255.255.224.0 c.255.255.255.240
28. Find the subnet work address for the following
IP address : 125.34.12.56 Mask 255.255.0.0
29. Find the subnet work address for the following:
IP address : 120.14.22.16 Mask : 255.255.128.0
30. Find the subnet work address for the following:
Ip address 140.11.36.22 Mask : 255.255.255.0