Question 1 - What is fragmentation? Explain its significance.

When an IP datagram travels from one host to another, it pass through different physical networks. 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). 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. 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 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. 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. Transport Layer: The Transport layer (also known as the Host-to-Host Transport layer) provides the Application layer with session and datagram communication services. The Transport layer encompasses the responsibilities of the OSI Transport layer. The core protocols of the Transport layer are TCP and UDP. TCP provides a oneto-one, connection-oriented, reliable communications service. TCP establishes connections, sequences and acknowledges packets sent, and recovers packets lost during transmission. In contrast to TCP, UDP provides a one-to-one or one-to-many, connectionless, unreliable communications service. UDP is used when the amount of data to be transferred is small (such as the data that would fit into a single packet), when an application developer does not want the overhead associated with TCP connections, or when the applications or upper-layer protocols provide reliable delivery. TCP and UDP operate over both IPv4 and IPv6 Internet layers. Question 3 - What is CIDR? Explain. Standard IP routing understands only Class A, B, and C network addresses. Wclass of the network number (thus the term classless). It is based solely on the high order bits of the IP address. These bits are called the IP prefix. Each CIDR routing table entry contains a 32-bit IP address and a 32-bit network mask, which together give the length and value of the IP prefix. This is represented as the tuple <IP_address network_mask>. However, there is no way to specify that multiple Class C networks are related. The result of this is termed the routing table explosion problem: A Class B network of 3000 hosts requires one routing table entry at each backbone router. The same environment, if addressed as a range of Class C networks, requires 16 entries. The solution to this problem is called Classless Inter-Domain Routing (CIDR). CIDR does not route according to the class of the network number (thus the term classless). It is based solely on the high order bits of the IP address. These bits are called the IP prefix. Each CIDR routing table entry contains a 32-bit IP address and a 32-bit network mask, which together give the length and value of the IP prefix. This is represented as the tuple <IP_address network_mask>. For example, to address a block of eight Class C addresses with one single routing table entry, the following representation suffices: <192.32.136.0 255.255.248.0>. This refers, from a backbone point of view, to the Class C network range from 192.32.136.0 to 192.32.143.0 as one single network. This is illustrated in Figure 1.

Figure 1: Classless Inter-Domain Routing: IP supernetting example This process of combining multiple networks into a single entry is referred to as Supernetting. Routing is based on network masks that are shorter than the natural network mask of an IP address. This contrasts with subnetting where the subnet masks are longer than the natural network mask. Question 4 - What is congestion? Mention few algorithms to overcome congestion. Congestion occurs when a link or node is carrying so much data that its quality of service deteriorates. Typical effects include queueing delay, packet loss or the blocking of new connections. A consequence of these latter two is that incremental increases in offered load lead either only to small increases in network throughput, or to an actual reduction in network throughput. Following are the few algorithms to overcome congestion: i) Slow Start: Old implementations of TCP start a connection with the sender injecting multiple segments into the network, up to the window size advertised by the receiver. Although this is OK when the two hosts are on the same LAN, if there are routers and slower links between the sender and the receiver, problems can arise. The algorithm to avoid this is called slow start. It operates by observing that the rate at which new packets should be injected into the network is the rate at which the acknowledgments are returned by the other end. Slow start adds another window to the sender's TCP: the congestion window, called cwnd. The sender starts by transmitting one segment and waiting for its ACK. When that ACK is received, the congestion window is incremented from one to two, and two segments can be sent. When each of those two segments is acknowledged, the congestion window is increased to four. At some point, the capacity of the IP network (for example, slower WAN links) can be reached, and an intermediate router will start discarding packets. This tells the sender that its congestion window has gotten too large. See Fig. 2 for an overview of slow start in action.

Fig. 2: TCP: Slow start in action ii) Congestion Avoidance: The assumption of the algorithm is that packet loss caused by damage is very small (much less than 1%). Therefore, the loss of a packet signals congestion somewhere in the network between the source and destination. There are two indications of packet loss:

A timeout occurs. Duplicate ACKs are received.

Congestion avoidance and slow start are independent algorithms with different objectives. But when congestion occurs, TCP must slow down its transmission rate of packets into the network and invoke slow start to get things going again. In practice, they are implemented together. Congestion avoidance and slow start require that two variables be maintained for each connection: A congestion window, cwnd A slow start threshold size, ssthresh

iii) Fast Retransmit: Fast retransmit avoids having TCP wait for a timeout to resend lost segments. Before describing the change, realize that TCP can generate an immediate acknowledgment (a duplicate ACK) when an out-of-order segment is received. This duplicate ACK should not be delayed. The purpose of this duplicate ACK is to let the other end know that a segment was received out of order and to tell it what sequence number is expected. iv) Fast recovery: After fast retransmit sends what appears to be the missing segment, congestion avoidance, but not slow start, is performed. This is the fast recovery algorithm. It is an improvement that allows high throughput under moderate congestion, especially for large windows. The reason for not performing slow start in this case is that the receipt of the duplicate ACKs tells TCP more than just a packet has been lost. Because the receiver can only generate the duplicate ACK when another segment is received, that segment has left the network and is in the receiver's buffer.

Question 5 - Explain the following with respect to Transport Protocols: a. User Datagram Protocol (UDP) b. Transmission Control Protocol (TCP) a. User Datagram Protocol (UDP): UDP is a standard protocol with STD number 6. Its status is standard and almost every TCP/IP implementation intended for small data units transfer or those which can afford to lose a little amount of data (such as multimedia streaming) will include UDP. It adds no reliability, flow-control or error recovery to IP. It simply serves as a multiplexer/de-multiplexer for sending and receiving datagrams, using ports to direct the datagrams, as shown in Fig.3 for a more detailed discussion of ports.

Fig. 3: UDP: De-Multiplexing based on ports UDP provides a mechanism for one application to send a datagram to another. The UDP layer can be regarded as being extremely thin and is, consequently, very efficient, but it requires the application to take responsibility for error recovery and so on. b. Transmission Control Protocol (TCP): TCP is a standard protocol with STD number 7. TCP is described by RFC 793 Transmission Control Protocol. Its status is standard, and in practice, every TCP/IP implementation that is not used exclusively for routing will include TCP. TCP provides considerably more facilities for applications than UDP. Specifically, this includes error recovery, flow control, and reliability. TCP is a connection-oriented protocol, unlike UDP, which is connectionless. Most of the user application protocols, such as Telnet and FTP, use TCP. The two processes communicate with each other over a TCP connection (Inter-Process Communication, or IPC), as shown in Fig. 3.1. In the figure, processes 1 and 2 communicate over a TCP connection carried by IP datagrams.

Fig. 3.1: TCP: Connection between Processes 6. With diagram explain the components of a VoIP networking system. Figure illustrates the main components of a VoIP networking system.

Figure 4: VoIP architecture Softswitch In the traditional PSTN model, a switchboard was used to connect two separate phone circuits into one contiguous line. In the VoIP model, this can be done through software. This concept is the softswitch, and it controls connections between the circuit and packet networks in segments where end-to-end VoIP is not yet available. The softswitch can be broken into two entities: the Call Agent and the Media Gateway. Media Gateway The Media Gateway is used to connect different physical networks in order to provide end-to-end connectivity. It functions very similarly to a typical network switch in that it can create a heterogeneous link between endpoints, regardless of the network media in between. Call Agent The Call Agent serves several purposes. It can use a TCP/IP link to manage multiple Media Gateways. Primarily, however, it manages the following functions: Billing Call routing Network signaling Phone handset There are two different types of handsets: Softphone: The relationship between a traditional phone handset and a softphone is similar to the relationship between a switchboard and a softswitch. The functionality of the softphone is provided through software, often running on a personal computer. The hardware involved typically consists of any combination of computer sound cards, headphones, or speakers, and a microphone. This implementation is currently included in common instant message programs and is often used in PC-toPC calls. Hardphones: The hardphone and the traditional handset correspond almost one-to-one. The hardphone is simply a traditional telephone handset that is VoIP capable. Both types of VoIP phones are telephones capable of carrying voice traffic using the Session Initiation Protocol

Analog Terminal Adapters (ATA) Analog Terminal Adapters (ATA) perform analog and digital conversions between a traditional analog phone and the broadband modem, allowing users to use the VoIP service with an existing phone. Coder-decoder (CODEC) The processing of digitizing a voice sample, or converting a digitized signal into an analog signal, requires the use of a coder-decoder. Each softphone, hardphone, or ATA must implement a CODEC in order to implement VoIP. There is a set of international CODEC standards, any of which can be used to perform this translation. These standards include: G.711: The G.711 (64 kps) standard, which encodes non-compressed speech streams running at 64 Kbps. G.726: G.726 is an ITU-T ADPCM speech CODEC standard covering the transmission of voice at rates of 16, 24, 32, and 40 kbps. G.728: G.728 is a ITU-T standard for speech coding operating at 16 kbps. The technology used is LDCELP, Low Delay Code Excited Linear Prediction. G.723.1: G.723.1 is an audio CODEC for voice that compresses voice audio in chunks of 30 milliseconds. A look-ahead of 7.5 ms duration is also used. G.729: G.729 is an audio data compression algorithm for voice that compresses voice audio in chunks of 10 milliseconds. Music or tones such as DTMF or fax tones cannot be transported reliably with this CODEC, and thus uses G.711 or out-of-band methods to transport these signals.G.729 is mostly used in Voice over IP (VoIP) applications for its low bandwidth requirement. Core IP network Given the name core IP network, it would be easy to assume that this refers simply to the Internet between two endpoints. While this might be true to some extent, the core IP network generally refers to a subset of the Internet that can guarantee a quality of service when transporting a stream of IP packets between voice applications. In many cases, this subset is proprietary and made available by a specific ISP offering voice services. Signaling gateway In order to implement VoIP communications, the functionality to notify an endpoint that another endpoint desires communication is necessary (such as by causing the recipients phone to ring). This is called signalling. Operational support system OSS provides user account setup, VoIP service provision, and fulfillment. If there are any issues in VoIP services, OSS provides trouble ticketing, troubleshooting, and service assurance. OSS also provides billing and other operational supports.