Question 1- What is fragmentation?

Explain its significance Ans: Fragmentation is the process of breaking up a single Internet Protocol (IP) datagram into multiple packets of smaller size. Every network link has a characteristic size of messages that may be transmitted, called the maximum transmission unit (MTU). An un-fragmented 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: (i) 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. (ii) 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. (iii) 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 un-fragmented 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.

(iv) 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.

Question 2- Briefly discuss the functions of transport layer Ans: In computer networking, the transport layer or layer 4 provides end-to-end communication services for applications within a layered architecture of network components and protocols. The transport layer provides convenient services such as connection-oriented data stream support, reliability, flow control, and multiplexing. There are following function of transport layer: Connection-oriented communication: It is normally easier for an application to interpret a connection as a data stream rather than having to deal with the underlying connection-less models, such as the datagram model of the User Datagram Protocol (UDP) and of the Internet Protocol (IP). Byte orientation: Rather than processing the messages in the underlying communication system format, it is often easier for an application to process the data stream as a sequence of bytes. This simplification helps applications work with various underlying message formats. Same order delivery: The network layer doesn't generally guarantee that packets of data will arrive in the same order that they were sent, but often this is a desirable feature. This is usually done through the use of segment numbering, with the receiver passing them to the application in order. This can cause head-of-line blocking. Reliability: Packets may be lost during transport due to network congestion and errors. By means of an error detection code, such as a checksum, the transport protocol may check that the data is not corrupted, and verify correct receipt by sending an ACK or NACK message to the sender. Flow control: The rate of data transmission between two nodes must sometimes be managed to prevent a fast sender from transmitting more data than can be supported by the receiving data buffer, causing a buffer overrun.

Question 2- What is CIDR? Explain. Ans: Classless Inter-Domain Routing (CIDR) is a method for allocating IP addresses and routing Internet Protocol packets. The Internet Engineering Task Force introduced CIDR in 1993 to replace the previous addressing architecture of

classful network design in the Internet. Their goal was to slow the growth of routing tables on routers across the Internet, and to help slow the rapid exhaustion of IPv4 addresses. IP addresses are described as consisting of two groups of bits in the address: the most significant part is the network address which identifies a whole network or subnet and the least significant portion is the host identifier, which specifies a particular host interface on that network. This division is used as the basis of traffic routing between IP networks and for address allocation policies. Classful network design for IPv4 sized the network address as one or more 8-bit groups, resulting in the blocks of Class A, B, or C addresses. Classless Inter-Domain Routing allocates address space to Internet service providers and end users on any address bit boundary, instead of on 8-bit segments. In IPv6, however, the interface identifier has a fixed size of 64 bits by convention, and smaller subnets are never allocated to end users.

CIDR notation is a syntax of specifying IP addresses and their associated routing prefix. It appends to the address a slash character and the decimal number of leading bits of the routing prefix, e.g., 192.168.0.0/16 for IPv4, and 2001:db8::/32 for IPv6. Question 4- What is congestion? Mention few algorithms to overcome congestion. Congestion, in the context of networks, refers to a network state where a node or link carries so much data that it may deteriorate network service quality, resulting in queuing delay, frame or data packet loss and the blocking of new connections. In a congested network, response time slows with reduced network throughput. Congestion occurs when bandwidth is insufficient and network data traffic exceeds capacity. Data packet loss from congestion is partially countered by aggressive network protocol retransmission, which maintains a network congestion state after reducing the initial data load. This can create two stable states under the same data traffic load - one dealing with the initial load and the other maintaining reduced network throughput. Cause When more packets were sent than could be handled by intermediate routers, the intermediate routers discarded many packets, expecting the end points of the network to retransmit the information. However, early TCP implementations had very bad retransmission behaviour. When this packet loss occurred, the end points sent extra packets that repeated the information lost, doubling the data rate sent, exactly the opposite of what should be done during congestion. This pushed the entire network into a 'congestion collapse' where most packets were lost and the resultant throughput was negligible. Congestion control Congestion control concerns controlling traffic entry into a telecommunications network, so as to avoid congestive collapse by attempting to avoid

oversubscription of any of the processing or link capabilities of the intermediate nodes and networks and taking resource reducing steps, such as reducing the rate of sending packets. It should not be confused with flow control, which prevents the sender from overwhelming the receiver. Few algorithms to overcome congestion Transient behaviours evaluated in the paper Equation-Based Congestion Control Algorithm General AIMD-Based Congestion Control Algorithms Binomial Congestion Control Algorithm

Question 5- Explain the following with respect to Transport Protocols: a. User Datagram Protocol (UDP) (5) b. Transmission Control Protocol (TCP) (5) a. User Datagram Protocol (UDP) (5) Ans: Unlike TCP, UDP doesn't establish a connection before sending data, it just sends. Because of this, UDP is called "Connectionless". UDP packets are often called "Datagrams". An example of UDP in action is the DNS service. DNS servers send and receive DNS requests using UDP. In this section we have to look at User Datagram protocol. Its a transport layer protocol. This section will cover the UDP protocol, its header structure & the way with which it establishes the network connection. The User Datagram Protocol (UDP) is a transport layer protocol that supports Network Application. It layered on just below the Session and sits above the IP (Internet Protocol) in open system interconnection model (OSI). This protocol is similar to TCP (transmission control protocol) that is used in client/ server programs like video conference systems expect UDP is connection less. UDP is a connectionless and unreliable transport protocol. The two ports serve to identify the end points within the source and destination machines. User Datagram Protocol is used, in place of TCP, when a reliable delivery is not required. However, UDP is never used to send important data such as webpages, database information, etc. Streaming media such as video, audio and others use UDP because it offers speed. The reason UDP is faster than TCP is because there is no form of flow control. No error checking, error correction, or acknowledgment is done by UDP.UDP is only concerned with speed. So when, the data sent over the Internet is affected by collisions, and errors will be present. b. Transmission Control Protocol (TCP) (5) After going through the various layers of the Model, its a time to have a look at the TCP protocol and to study its functionality. This section will help the reader to get know about the concepts of the TCP, characteristics of the TCP and then would gradually take him into the details of TCP like connection

establishment/closing, communication in TCP and why the TCP protocol is called as reliable as well as adaptive protocol. This section will end with a comparison between UDP and TCP followed by a nice exercise which would encourage readers to solve more and more problems. In theory, a transport layer protocol could be a very simple software routine, but TCP protocol cannot be called simple. Why use a transport layer which is as complex as TCP? The most important reason depends on IP's unreliability. In fact all the layers below TCP are unreliable and deliver the datagram hop-byhop. IP layer delivers the datagram hop-by-hop and does not guarantee delivery of a datagram; it is a connectionless system. IP simply handles the routing of datagrams; and if problems occur, IP discards the packet without a second thought generating an error message back to the sender in the process. The task of ascertaining the status of the datagrams sent over a network and handling the resending of information if parts have been discarded falls to TCP TCP manages the flow of datagrams from the higher layers, as well as incoming datagrams from the IP layer. It has to ensure that priorities and security are respected. TCP must be capable of handling the termination of an application above it that was expecting incoming datagrams, as well as failures in the lower layers. TCP also must maintain a state table of all data streams in and out of the TCP layer. The isolation of these services in a separate layer enables applications to be designed without regard to flow control or message reliability. Without the TCP layer, each application would have to implement the services themselves, which is a waste of resources. Characteristics of TCP Unicast protocol: TCP is based on a unicast network model, and supports data exchange between precisely two parties. It does not support broadcast or multicast network models. Connection state: Rather than impose a state within the network to support the connection, TCP uses synchronized state between the two endpoints. Reliable : Reliability implies that the stream of octets passed to the TCP driver at one end of the connection will be transmitted across the network so that the stream is presented to the remote process as the same sequence of octets, in the same order as that generated by the sender. Full duplex: TCP is a full-duplex protocol; it allows both parties to send and receive data within the context of the single TCP connection. Streaming: Although TCP uses a packet structure for network transmission, TCP is a true streaming protocol, and application-level network operations are not transparent.

Question 6 - With diagram explain the components of a VoIP networking system. Ans: Definition: VoIP is a technology that allows telephone calls to be made over computer networks like the Internet. VoIP converts analog voice signals

into digital data packets and supports real-time, two-way transmission of conversations using Internet Protocol (IP). VoIP calls can be made on the Internet using a VoIP service provider and standard computer audio systems. Alternatively, some service providers support VoIP through ordinary telephones that use special adapters to connect to a home computer network. Many VoIP implementations are based on the H.323 technology standard. There are three main components of a VoIP network: user agents, gateways, and proxy servers.

User Agent In a VoIP network, any device that can make or receive telephone calls is called a User Agent (UA). Each User Agent contains a User Agent Server (UAS) responsible for handling requests from another endpoint, (for example, inbound calls) and a User Agent Client (UAC) which generates requests, (for example, outbound calls) for other endpoints. Examples of User Agent Clients and User Agent Servers are:

A SIP hard phone. A SIP soft phone. WebSphere Voice Response (which simulates a number of phones) for incoming or outgoing calls.

Gateways A gateway is a device which acts as a bridge between VoIP and the PSTN network. A gateway can take an incoming call from a T1 interface and convert the signalling into SIP message exchanges, and convert the voice from TDM into RTP packets. Proxy servers In a SIP system, a proxy server (used with a registrar and a location server), can provide the following services:

Call Routing including URI translation. Registration.

Access (authentication) to a SIP network.

A Proxy server is the means by which calls are routed within a SIP VoIP network. For example, a telephony gateway might be configured to send all incoming calls to the SIP proxy server which will then route the calls to specific endpoints (this can include load balancing or skills-based routing).