MC0087- Internetworking with TCP/IP

(Book ID: B1008)

Q1. What is fragmentation? Explain its significance. Ans: Maximum Transmission Unit (MTU):- MTU is the maximum frame size of physical network. During propagation of IP datagram, it can pass through different physical networks. MTUlimits 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 unfragmented 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 be discarded 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 datagrams are minor modifications 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.


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4. Each of these fragmented datagrams is now forwarded as a normal IP datagram. IP handles each fragment 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 Q2. Briefly discuss the functions of transport layer Ans: Functions of transport layers are:Accepts data from session layer breaks it into packets and delivers these packets to thenetwork layer. Guarantee successful arrival of data at thedestination device. Provide end-to-end dialog that is the transport layer at the source device directlycommunicates with transport layer at destination device. Message headers and control messages are usedfor this purpose. Separates the upper layers from the low level details of data transmission and makessure an efficient delivery. OSI model provides connection-oriented service at transport layer. 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 thetransmission of isolated messages. There is no surety that these isolated messages are delivered to thedestination devices in case of broadcast networks and they will be in the same order as were sent from thesource. If the network layer do not provide adequate services for the data transmission. Data loss due to poornetwork management is handled by using transport layer. Examine packets that are lost ordamaged along the way.


Q3. What is CIDR? Explain
Ans:CIDR (Classless Inter-Domain Routing, sometimes known assuper netting) is a way toallocate and specify the Internet addresses used in inter-domain routing more flexibly than withthe original system of Internet Protocol (IP) address classes. As a result, the number of availableInternet addresses has been greatly increased. CIDR is now the routing system used by virtuallyall gateway hosts on the Internet's backbone network. The Internet's regulating authorities nowexpect 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 addressformat to a network address and the remaining portion to the specific host machines within thenetwork 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 254host machines but far fewer than the 65,533 host addresses possible would essentially be"wasting" MC0087 Page 2

most of the block of addresses allocated. For this reason, the Internet was, until thearrival of CIDR, running out of address space much more quickly than necessary. CIDReffectively 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 ispossible, greatly expanding the number of possible addresses on the Internet. However, it will besome time before IPv6 is in widespread use.)Using CIDR, each IP address has anetwork prefixthat 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 anyarbitrary class assignment structure). A destination IP address or route that describes manypossible destinations has a shorter prefix and is said to be less specific. A longer prefix describesa destination gateway more specifically. Routers are required to use the most specific or longestnetwork prefix in the routing table when forwarding packets. A CIDR network address looks like this: The "" is the network address itself and the "18" says that the first 18 bits are thenetwork part of the address, leaving the last 14 bits for specific host addresses. CIDR lets onerouting table entry represent an aggregation of networks that exist in the forward path that don'tneed to be specified on that particular gateway, much as the public telephone system uses areacodes to channel calls toward a certain part of the network. This aggregation of networks in asingle address is sometimes referred to as asupernet. CIDR is supported by the Border Gateway Protocol, the prevailing exterior (interdomain)gateway protocol. (The older exterior or interdomain gateway protocols, Exterior GatewayProtocol and Routing Information Protocol, do not support CIDR.) CIDR is also supported by theOSPF interior or intradomain gateway protocol. Block of subnetted into 32 subnets: a) Given block /24 have 256 addresses (0-255). Divide 256 by 32 to determine that each subnetwill have 8 addresses. Using binary, determine the net mask that will achieve a total of 8addresses in each network. Since 0 is the first value in a range of 8 addresses, we want zerothrough 7 or (000-111). This confirms that only 3 bits are required to represent the addresses ineach of the 32 subnets. (xxxxxyyy - xxxxxyyy) illustrates the binary range for the addresses within each of the 32subnets. 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 netmask is /24 + /5 = /29 or 11111111.11111111.11111111.11111000 = 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) xxxxxyyy - represents the bits used to define the network and host portion of the IP address. 00000yyy = first subnet in range 00001yyy = second subnet in range 00010yyy = third subnet in range 00011yyy = fourth subnet in range MC0087 Page 3

11100yyy = 29th subnet in range 11101yyy = 30th subnet in range 11110yyy = 31st subnet in range 11111yyy = 32nd subnet in range The first and last addresses in subnet 1: 00000yyy = first subnet in range 00000000 = first address in subnet 1 (decimal 0) 00000111 = last address in subnet 1 (decimal 7) The first and last addresses in subnet 32: 11111yyy = 32nd subnet in range 11111000 = first address in subnet 32 (decimal 248) 11111111 = last address in subnet 32 (decimal 255)

Q4. What is congestion? Mention few algorithms to overcome congestion
Ans:TCP is the popular transport protocol for best-effort traffic in Internet. However, TCP is not well-suited for many applications such as streaming multimedia, because TCP congestion controlalgorithms introduce large variations in the congestion window size (and corresponding largevariations in the sending rate). Such variability in the sending rate is not acceptable to manymultimedia applications. Hence, many multimedia applications are built over UDP and use nocongestion control at all. The absence of congestion control in applications built over UDP maylead to congestion collapse on the Internet. In addition, the UDP flows may starve any competingTCP flows. To overcome these adverse effects, congestion control needs to be incorporated intoall applications using the Internet, whether at the transport layer or provided by the applicationitself. Furthermore, the congestion control algorithms must be TCP-friendly, i.e. the TCP-friendlyflows should not gain more throughput than competing TCP flows in the long run. Thus, in recentyears, many researchers have focussed on developing “TCP-friendly” transport protocols whichare suitable for many applications that currently use UDP. In this direction, IETF is currentlyworking on developing a new protocol called, Datagram Congestion Control Protocol (DCCP),that provides an unreliable datagram service with congestion control. DCCP is designed to useany suitable TCP- friendly congestion control algorithm. With a multitude of TCP-friendlycongestion control algorithms available, some important questions that need to be answered are:What are the strengths and weakness of the various TCP-friendly algorithms? Is there a singlealgorithm which is uniformly superior over other algorithms?. The first step in answering thesequestions is to study the short-term and long-term behavior of these algorithms. Although the goalof all TCP-friendly algorithms is to emulate the behavior of TCP in the long term, thesealgorithms may have an adverse impact in the short-term on competing TCP flows. Since TCP-friendly algorithms are designed for smoother sending rates than TCP, these algorithms may reactslowly to new connections that share a common bottleneck link. Such a slower response mayhave a deleterious effect on TCP flows. Forexample, a TCP connection suffering losses in itsslow 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 adetailed study is required on the short-term (transient)behavior of TCPfriendly flows in additionto their long-term behavior. In this paper, we study the transient behavior of three TCP-friendlycongestion control algorithms: general AIMD congestion control, TFRC and binomial congestioncontrol algorithm. Prior work has studied the transient behavior of these algorithms when REDqueues are used at the bottleneck link. However, as droptail MC0087 Page 4

queues are still widely used inpractice, in this paper we study the transient behavior of these algorithms with droptail queues. Past work has also identified certain unfairness of AIMD and binomial congestion controlalgorithms to TCP with droptail queues, but has not identified the reasons for this unfairness. In this paper, we analyze the reasons for this unfairness, and validate the analysis by simulations.The rest of the paper is organized as follows. In Section II, we briefly overview the various TCP-friendly congestioncontrol algorithms proposed in literature. In Section III, we define thetransientbehaviors studied in this paper, and analyze the expected transient behaviors of the various TCP-friendly congestion control algorithms. Section IV analyzes in detail the reasons for unfairness of AIMD and binomial congestion control algorithms with droptail queues. We present our simulation results 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 Algorithm C. General AIMD-Based Congestion Control Algorithms D. Binomial Congestion Control Algorithm Q5. Explain the following with respect to Transport Protocols: a. User Datagram Protocol (UDP) (5) b. Transmission Control Protocol (TCP) (5) Ans: User Datagram Protocol (UDP) : The User Datagram Protocol (UDP) is a transport layer protocoldefined for use with the IP network layer protocol. It is defined by RFC 768 written by JohnPostel. It provides a besteffort 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 IntermediateSystem (IS)). The simplicity of UDP reduces the overhead from using the protocol and theservices may be adequate 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 areunique in that they do not establish end-to-end connections between communicating end systems.UDP communication consequently does not incur connection establishment and teardownoverheads and there is minimal associated end system state. Because of these characteristics, UDPcan offer a very efficient communication transport to some applications, but has no inherentcongestion control or reliability. A second unique characteristic of UDP is that it provides noinherent On many platforms, applications can send UDP datagrams at the line rate of the link interface, which is often much greater than the available path capacity, and doing so wouldcontribute to congestion along the path, applications therefore need to be designed responsibly.One increasingly popular use of UDP is as a tunneling protocol, where a tunnel endpointencapsulates the packets of another protocol inside UDP datagrams and transmits them to anothertunnel endpoint, which decapsulates the UDP datagrams and forwards the original packetscontained in the payload. Tunnels establish virtual links that appear to directly connect locationsthat are distant in the physical Internet topology, MC0087 Page 5

and can be used to create virtual (private)networks. Using UDP as a tunneling protocol is attractive when the payload protocol is not supported by middleboxes that may exist along the path, because many middleboxes support UDPtransmissions.UDP does not provide any communications security. Applications that need to protect theircommunications against eavesdropping, tampering, or message forgery therefore need toseparately provide security services using additional protocol mechanisms. Transmission Control Protocol (TCP): The Transmission Control Protocol (TCP) is a connection-oriented reliable protocol. It provides a reliable transport service between pairs of processesexecuting on End Systems (ES) using the network layer service provided by the IP protocol. 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 andtransmitted by 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, whichonly provides a Best Effort service. To implement the service, TCP uses a number of protocoltimers that ensure reliable and synchronized 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 headercontains a 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 theseservices generally have the same value as those used for UDP services (a full list of all portnumbers is provided in the reference at the end of this page).

Q6. With diagram explain the components of a VoIP networking system.
Ans: IP Telephony Server(s)– This is the heart of the IP Telephony systems which providescomplete Call Control, Dial Plan control and all the basic voice applications (In case of smaller systems, all the functionalities of the below mentioned application servers can alsobe bundled with this) Application Servers–Some times applications like IVR (Interactive Voice Response–Auto Attendant), Call Recording, Voice Mail, Data Base Integration require to be hostedin separate servers - Especially for larger VOIP installations. IP Phones–These IP Phones connect directly to the IP Network (RJ-45 based UTPCables) and provide all the voice functionalities hitherto provided by analog phones likecaller ID display, speaker phones, speed dial keys, memory etc. Soft Phones–These are basically software utilities that have all the telephony functionsbut use the 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 comewith Wi-Fi adaptors and can be used as a Wi-Fi Phone (if the manufacturer supports thesame). Cell Phones can also connect to the IP Telephony server through 3G Networks/ CDMA networks for making a VOIP Call. MC0087 Page 6

Analog Telephony Adapters (ATA) (ATA)–These are specialized devices that connect to theLAN at one end and connect to FXO (Analog Trunks) or FXS (Analog Extensions) at theother end. PRI Cards–These These are used to connect PRI/E1/T1 Trunk Lines to IP Telephony Servers– Servers Usually they connect directly with the PCI/ PCI Express Slot in the server. Computer IP Network–An An IP based Computer Network is used t to o carry the voicesignals across 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 thenetwork. IP Networks need to be built with sufficient redundancy and security for continuousavailability of IP Telephony services–If If there is a DOS attack on the network (forexample), the telephones also become inactive along with the computers. Scaling of IP Telephony systems needs to be planned properly properly–Failing Failing which, the IPtelephony server may not be able to handle high concurrent call loads. There are hardware/ hardware license based restrictions on the maximum number of concurrent calls that a single servercan handle/ maximum number of end points that can connect to a single server.


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