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Summer 2013

Master of Computer Application (MCA) Semester 6

MC0087 Internetworking with TCP/IP 4 Credits

(Book ID: B1008)

Question. 1.:- What is fragmentation? Explain its significance..

Answer. :- When a data packet travels from one host to another, it can pass through different physicalnetworks. Each physical network has a maximum frame size. This is called the maximum transmission

unit (MTU). It limits the length of a datagram that can be placed in one physical frame. IP implements a

process to fragment datagrams exceeding the MTU. The process creates a set of datagrams within the

maximum size. The receiving host reassembles the original datagram. IP requires that each link support a

minimum MTU of 68 octets. This is the sum of the maximum IP header length (60 octets) and the

minimum possible length of data in a non-final fragment (8 octets). If any network provides a lower value

than this, fragmentation and reassembly must be implemented in the network interface layer. This must

be transparent to IP. IP implementations are not required to handle unfragmented datagrams larger than

576 bytes. In practice, most implementations will accommodate larger values. An unregimented datagram

has an all-zero fragmentation information field. That is, the more fragments flag bit is zero and the

fragment offset is zero. The following steps fragment the datagram:

1. The DF flag bit is checked to see if fragmentation is allowed. If the bit is set, the datagram will bediscarded and an ICMP error returned to the originator.2. Based on the MTU value, the data field is split into two or more parts. All newly created data portions

must have a length that is a multiple of 8 octets, with the exception of the last data portion.

3. Each data portion is placed in an IP datagram. The headers of these datagram are minormodifications of the original:

The more fragments flag bit is set in all fragments except the last.

The fragment offset field in each is set to the location this data portion occupied in the original

datagram, relative to the beginning of the original unfragmented datagram. The offset is measured in

8-octet units.

If options were included in the original datagram, the high order bit of the option type byte

determines if this information is copied to all fragment datagrams or only the first datagram. For

example, source route options are copied in all fragments.

The header length field of the new datagram is set. The total length field of the new datagram is set The header checksum field is re-calculated.

4. Each of these fragmented datagrams is now forwarded as a normal IP datagram. IP handles eachfragment independently. The fragments can traverse different routers to the intended destination.

They can be subject to further fragmentation if they pass through networks specifying a smaller MTU.

At the destination host, the data is reassembled into the original datagram. The identification field set

by the sending host is used together with the source and destination IP addresses in the datagram.

Fragmentation does not alter this field. In order to reassemble the fragments, the receiving host

allocates a storage buffer when the first fragment arrives. The host also starts a timer. When

subsequent fragments of the datagram arrive, the data is copied into the buffer storage at the location

indicated by the fragment offset field. When all fragments have arrived, the complete original

unfragmented datagram is restored. Processing continues as for unfragmented datagrams. If the

timer is exceeded and fragments remain outstanding, the datagram is discarded. The initial value of

this timer is called the IP datagram time to live (TTL) value. It is implementation-dependent. Some

implementations allow it to be configured. The netstat command can be used on some IP hosts to

list the details of fragmentation.

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Question. 2.- Briefly discuss the functions of transport layer..

Answer. :- Transport layer accepts data from session layer breaks it into packets and delivers these

packets to the network layer. It is the responsibility of transport layer to guarantee successful arrival of

data at the destination device. It provides an end-to-end dialog that is the transport layer at the source

device directly communicates with transport layer at destination device. Message headers and control

messages are used for this purpose. It separates the upper layers from the low level details of data

transmission and makes sure an efficient delivery. OSI model provides connection-oriented service at

transport layer.

It is responsible for the determination of the type of service that is to be provided to the upper

layer. Normally it transmits packets in the same order in which they are sent however it can also facilitate

the transmission of isolated messages. There is no surety that these isolated messages are delivered to

the destination devices in case of broadcast networks and they will be in the same order as were sent

from the source. If the network layer do not provide adequate services for the data transmission. Data

loss due to poor network management is handled by using transport layer. It checks for any packets that

are lost or damaged along the way.

The End !

Question. 3.- What is CIDR? Explain.

Answer. :- CIDR (Classless Inter-Domain Routing, sometimes known as super net t ing) is a way to

allocate and specify the Internet addresses used in inter-domain routing more flexibly than with the

original system of Internet Protocol (IP) address classes. As a result, the number of available Internet

addresses has been greatly increased. CIDR is now the routing system used by virtually all gateway

hosts on the Internet's backbone network. The Internet's regulating authorities now expect every Internet

service provider (ISP) to use it for routing.

The original Internet Protocol defines IP addresses in four major classes of address structure,

Classes A through D. Each of these classes allocates one portion of the 32-bit Internet address format to

a network address and the remaining portion to the specific host machines within the network specified by

the address. One of the most commonly used classes is (or was) Class B, which allocates space for up to

65,533 host addresses. A company who needed more than 254 host machines but far fewer than the

65,533 host addresses possible would essentially be "wasting" most of the block of addresses allocated.

For this reason, the Internet was, until the arrival of CIDR, running out of address space much more

quickly than necessary. CIDR effectively solved the problem by providing a new and more flexible way to

specify network addresses in routers. (With a new version of the Internet Protocol - IPv6 - a 128-bit

address is possible, greatly expanding the number of possible addresses on the Internet. However, it will

be some time before IPv6 is in widespread use.)

Using CIDR, each IP address has a network p ref ixthat identifies either an aggregation of

network gateways or an individual gateway. The length of the network prefix is also specified as part of

the IP address and varies depending on the number of bits that are needed (rather than any arbitrary

class assignment structure). A destination IP address or route that describes many possible destinations

has a shorter prefix and is said to be less specific. A longer prefix describes a destination gateway morespecifically. Routers are required to use the most specific or longest network prefix in the routing table

when forwarding packets CIDR network address looks like this: 192.30.250.00/18.The "192.30.250.00" is

the network address itself and the "18" says that the first 18 bits are the network part of the address,

leaving the last 14 bits for specific host addresses. CIDR lets one routing table entry represent an

aggregation of networks that exist in the forward path that don't need to be specified on that particular

gateway, much as the public telephone system uses area codes to channel calls toward a certain part of

the network. This aggregation of networks in a single address is sometimes referred to as a supernet.

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CIDR is supported by the Border Gateway Protocol, the prevailing exterior (interdomain) gateway

protocol. (The older exterior or interdomain gateway protocols, Exterior Gateway Protocol and Routing

Information Protocol, do not support CIDR.) CIDR is also supported by the OSPF interior or intradomain

gateway protocol. Block of 211.17.180.0/24 sub netted into 32 subnets:

A. Given block /24 have 256 addresses (0-255). Divide 256 by 32 to determine that each subnet will

have 8 addresses. Using binary, determine the net mask that will achieve a total of 8 addresses

in each network. Since 0 is the first value in a range of 8 addresses, we want zero through 7 or

(000-111). This confirms that only 3 bits are required to represent the addresses in each of the 32

subnets.

(xxxxxyyy - xxxxxyyy) illustrates the binary range for the addresses within each of the 32 subnets.

The "xxxxx" represents the additional 5 bits that will be used to define the network portion of the IP

address. The "yyy" portion illustrates the host portion of the IP address.

This means that the resulting net mask is /24 + /5 = /29 or 255.255.255.248

11111111.11111111.11111111.11111000 = 255.255.255.248

B. The subnet mask above defines that 3 bits are used to define the host portion of the IP address.

binary 111 = decimal 7 (considering range of 0-7, shows 8 addresses per subnet)

C. xxxxx yyy - represents the bits used to define the network and host portion of the IP address.00000yyy = first subnet in range

00001yyy = second subnet in range00010yyy = third subnet in range00011yyy = fourth subnet in range11100yyy = 29th subnet in range11101yyy = 30th subnet in range11110yyy = 31st subnet in range11111yyy = 32nd subnet in range.

The first and last addresses in subnet 1:

00000yyy = first subnet in range00000000 = first address in subnet 1 (decimal 0) 211.17.180.0/2900000111 = last address in subnet 1 (decimal 7) 211.17.180.7/29

The first and last addresses in subnet 32:

11111yyy = 32nd subnet in range11111000 = first address in subnet 32 (decimal 248) 211.17.180.248/2911111111 = last address in subnet 32 (decimal 255) 211.17.180.255/29

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Question. 4.:- what is congestion. Mention few algorithms to overcome congestion.

Answer. :- TCP is the popular transport protocol for best-effort traffic in Internet. However, TCP is notwell-suited for many applications such as streaming multimedia, because TCP congestion controlalgorithms introduce large variations in the congestion window size (and corresponding large variations inthe sending rate). Such variability in the sending rate is not acceptable to many multimedia applications.

Hence, many multimedia applications are built over UDP and use no congestion control at all. Theabsence of congestion control in applications built over UDP may lead to congestion collapse on theInternet. In addition, the UDP flows may starve any competing TCP flows. To overcome these adverseeffects, congestion control needs to be incorporated into all applications using the Internet, whether at thetransport layer or provided by the application itself. Furthermore, the congestion control algorithms mustbe TCP-friendly, i.e. the TCP-friendly flows should not gain more throughput than competing TCP flows inthe long run. Thus, in recent years, many researchers have focussed on developing TCP -friendlytransport protocols which are suitable for many applications that currently use UDP. In this direction, IETFis currently working on developing a new protocol called, Datagram Congestion Control Protocol (DCCP),that provides an unreliable datagram service with congestion control. DCCP is designed to use anysuitable TCP- friendly congestion control algorithm. With a multitude of TCP-friendly congestion controlalgo- rithms available, some important questions that need to be answered are: What are the strengthsand weakness of the various TCP-friendly algorithms? Is there a single algorithm which is uniformly

superior over other algorithms. The first step in answering these questions is to study the short-term andlong-term behavior of these algorithms. Although the goal of all TCP-friendly algorithms is to emulate thebehavior of TCP in the long term, these algorithms may have an adverse impact in the short-term oncompeting TCP flows. Since TCP-friendly algorithms are designed for smoother sending rates than TCP,these algorithms may react slowly to new connections that share a common bottleneck link. Such aslower response may have a deleterious effect on TCP flows. For example, a TCP connection sufferinglosses in its slow start phase may enter the congestion avoidance phase with a small window, andconsequently obtain lesser throughput than other competing flows. Hence, it is clear that a detailed studyis required on the short-term (transient)behavior of TCP-friendly flows in addition to their long-termbehavior. In this paper, we study the transient behavior of three TCP-friendly congestion controlalgorithms: general AIMD congestion control, TFRC and binomial congestion control algorithm . Priorwork has studied the transient behavior of these algorithms when RED queues are used at the bottlenecklink. However, as droptail queues are still widely used in practice, in this paper we study the transient

behavior of these algorithms with droptail queues. Past work has also identified certain unfairness ofAIMD and binomial congestion control algorithms to TCP with droptail queues, but has not identified thereasons for this unfairness. In this paper, we analyze the reasons for this unfairness, and validate theanalysis by simulations. The rest of the paper is organized as follows. In Section II, we briefly overviewthe various TCP-friendly congestion control algorithms proposed in literature. In Section III, we define thetransient behaviors studied in this paper, and analyze the expected transient behaviors of the variousTCP-friendly congestion control algorithms. Section IV analyzes in detail the reasons for unfairness ofAIMD and binomial congestion control algorithms with droptail queues. We present our sim- ulationresults in Section V, and we conclude in Section VI.

few algorithms to overcome congestion

A. Transient behaviors evaluated in the paper

B. Equation-Based Congestion Control AlgorithmC. General AIMD-Based Congestion Control AlgorithmsD. Binomial Congestion Control Algorithm

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Question. 5 :- Explain the following with respect to Transport Protocols:

A. User Datagram Protocol (UDP)B. Transmission Control Protocol (TCP)

Answer. :-

A.:- User Datagram Protocol (UDP) : The User Datagram Protocol (UDP) is a transport layer

protocol defined for use with the IP network layer protocol. It is defined by RFC 768 written by JohnPostel. It provides a best-effort datagram service to an End System (IP host).

The service provided by UDP is an unreliable service that provides no guarantees for delivery andno protection from duplication (e.g. if this arises due to software errors within an Intermediate System(IS)). The simplicity of UDP reduces the overhead from using the protocol and the services may beadequate in many cases.

UDP provides a minimal, unreliable, best-effort, message-passing transport to applications andupper-layer protocols. Compared to other transport protocols, UDP and its UDP-Lite variant are unique inthat they do not establish end-to-end connections between communicating end systems. UDPcommunication consequently does not incur connection establishment and teardown overheads andthere is minimal associated end system state. Because of these characteristics, UDP can offer a veryefficient communication transport to some applications, but has no inherent congestion control orreliability. A second unique characteristic of UDP is that it provides no inherent On many platforms,applications can send UDP datagrams at the line rate of the link interface, which is often much greaterthan the available path capacity, and doing so would contribute to congestion along the path, applicationstherefore need to be designed responsibly. One increasingly popular use of UDP is as a tunnelingprotocol, where a tunnel endpoint encapsulates the packets of another protocol inside UDP datagramsand transmits them to another tunnel endpoint, which decapsulates the UDP datagrams and forwards theoriginal packets contained in the payload. Tunnels establish virtual links that appear to directly connectlocations that are distant in the physical Internet topology, and can be used to create virtual (private)networks. Using UDP as a tunneling protocol is attractive when the payload protocol is not supported bymiddle boxes that may exist along the path, because many middle boxes support UDP transmissions.

UDP does not provide any communications security. Applications that need to protect their

communications against eavesdropping, tampering, or message forgery therefore need to separatelyprovide security services using additional protocol mechanisms.

Protocol Header

A computer may send UDP packets without first establishing a connection to the recipient. A UDPdatagram is carried in a single IP packet and is hence limited to a maximum payload of 65,507 bytes forIPv4 and 65,527 bytes for IPv6. The transmission of large IP packets usually requires IP fragmentation.Fragmentation decreases communication reliability and efficiency and should therefore be avoided.

To transmit a UDP datagram, a computer completes the appropriate fields in the UDP header (PCI) andforwards the data together with the header for transmission by the IP network layer.

Th eUDP protoco l header consis ts of 8 bytes of Protoco l Contro l In format ion (PCI)

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TCP is stream oriented, that is, TCP protocol entities exchange streams of data. Individual bytesof data 9e.g. from an application or session layer protocol) are placed in memory buffers and transmittedby TCP in transport Protocol Data Units (for TCP these are usually known as "segments"). The reliable,flow-controlled TCP service is much more complex than UDP, which only provides a Best Effort service.

To implement the service, TCP uses a number of protocol timers that ensure reliable andsynchronized communication between the two End Systems.

For most networks approximately 90% of current traffic uses this transport service. It is used bysuch applications as telnet, World Wide Web (WWW), ftp, electronic mail. The transport header containsa Service Access Point which indicates the protocol which is being used (e.g. 23 = Telnet; 25 = Mail; 69 =TFTP; 80 = WWW (http)). The port numbers associated with these services generally have the samevalue as those used for UDP services (a full list of all port numbers is provided in the reference at the endof this page).

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Question. 6 :- With diagram explain the components of a VoIP networking system.

Answer :- Diagram of a VoIP Networking System.:-

Figure Of VoIP networking system.

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The components of a VoIP networking system :-

IP Telephony Server(s) This is the heart of the IP Telephony systems which provides completeCall Control, Dial Plan control and all the basic voice applications (In case of smaller systems, allthe functionalities of the below mentioned application servers can also be bundled with this)

Application Servers Some times applications like IVR (Interactive Voice Response AutoAttendant), Call Recording, Voice Mail, Data Base Integration require to be hosted in separateservers Especially for larger VOIP installations.

IP Phones These IP Phones connect directly to the IP Network (RJ-45 based UTP Cables) andprovide all the voice functionalities hitherto provided by analog phones like caller ID display,speaker phones, speed dial keys, memory etc.

Soft Phones These are basically software utilities that have all the telephony functions but usethe computer, head-set with microphone to make and receive calls.

Wi-Fi Phones/ Dual Mode Cell Phones Wi-Fi phones are based on IP Technology andconnect to the wireless network and act as mobile extensions. Certain Cell phones come with Wi-Fi adaptors and can be used as a Wi-Fi Phone (if the manufacturer supports the same). CellPhones can also connect to the IP Telephony server through 3G Networks/ CDMA networks formaking a VOIP Call.

Analog Telephony Adapters (ATA) These are specialized devices that connect to the LAN at

one end and connect to FXO (Analog Trunks) or FXS (Analog Extensions) at the other end. PRI Cards These are used to connect PRI/E1/T1 Trunk Lines to IP Telephony Servers

Usually they connect directly with the PCI/ PCI Express Slot in the server.

Computer IP Network An IP based Computer Network is used to carry the voice signalsacross the enterprise and sometimes even to remote locations.

IP Phones are much more expensive when compared to the cost of analog phones.

The voice call quality (over IP Networks) depends on a number of parameters like theconfiguration of right QoS parameters, latency, jitter, available bandwidth etc across the network.

IP Networks need to be built with sufficient redundancy and security for continuous availability ofIP Telephony services If there is a DOS attack on the network (for example), the telephonesalso become inactive along with the computers.

Scaling of IP Telephony systems needs to be planned properly Failing which, the IP telephonyserver may not be able to handle high concurrent call loads. There are hardware/ license basedrestrictions on the maximum number of concurrent calls that a single server can handle/maximum number of end points that can connect to a single server.

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