Network Layer-I PDF

Network Layer-I: The network layer in the TCP/IP protocol suite is responsible for the host-to-host delivery of datagrams. It provides services to the transport layer and receives services from the data-link layer. A datagram is a basic transfer unit associated with a packet-switched network. Datag...

Network Layer-I: The network layer in the TCP/IP protocol suite is responsible for the host-to-host delivery of datagrams. It provides services to the transport layer and receives services from the data-link layer. A datagram is a basic transfer unit associated with a packet-switched network. Datagrams are typically structured in header and payload sections. Datagrams provide a connectionless communication service across a packet-switched network. NETWORK-LAYER SERVICES Before discussing the network layer in the Internet today, let’s briefly discuss the  Network Layer-I: The figure shows that the Internet is made of many networks (or links) connected through the connecting devices. In other words, the Internet is an internetwork, a combination of LANs and WANs. To better understand the role of the network layer (or the internetwork layer), we need to think about the connecting devices (routers or switches) that connect the LANs and WANs. As the figure shows, the network layer is involved at the source host, destination host, and all routers in the path (R2, R4, R5, and R7). At the source host (Alice), the network layer accepts a packet from a transport layer, encapsulates the packet in a datagram, and delivers the packet to the data-link layer. At the destination host (Bob), the datagram is decapsulated, and the packet is extracted and delivered to the corresponding transport layer.  Network Layer-I: Although the source and destination hosts are involved in all five layers of the TCP/IP suite, the routers use three layers if they are routing packets only; however, they may need the transport and application layers for control purposes. A router in the path is normally shown with two data-link layers and two physical layers, because it receives a packet from one network and delivers it to another network. Packetizing The first duty of the network layer is definitely packetizing: encapsulating the payload (data received from upper layer) in a network-layer packet at the source and decapsulating the payload from the network-layer packet at the destination. The source host receives the payload from an upper-layer protocol, adds a header that contains the source and destination addresses and some other information that is required by the network-layer protocol (as discussed later) and delivers the packet to the data-link layer.  Network Layer-I: The source is not allowed to change the content of the payload unless it is too large for delivery and needs to be fragmented. The destination host receives the network-layer packet from its data- link layer, decapsulates the packet, and delivers the payload to the corresponding upper-layer protocol. If the packet is fragmented at the source or at routers along the path, the network layer is responsible for waiting until all fragments arrive, reassembling them, and delivering them to the upper-layer protocol. The routers in the path are not allowed to decapsulate the packets they received unless the packets need to be fragmented. The routers are not allowed to change source and destination addresses either.  Network Layer-I: Routing and Forwarding Routing The network layer is responsible for routing the packet from its source to the destination. A physical network is a combination of networks (LANs and WANs) and routers that connect them. This means that there is more than one route from the source to the destination. The network layer is responsible for finding the best one among these possible routes. The network layer needs to have some specific strategies for defining the best route. In the Internet today, this is done by running some routing protocols to help the routers coordinate their knowledge about the neighborhood and to come up with consistent tables to be used when a packet arrives.  Network Layer-I: Forwarding forwarding can be defined as the action applied by each router when a packet arrives at one of its interfaces. The decision-making table a router normally uses for applying this action is sometimes called the forwarding table and sometimes the routing table. When a router receives a packet from one of its attached networks, it needs to forward the packet to another attached network (in unicast routing) or to some attached networks (in multicast routing). To make this decision, the router uses a piece of information in the packet header, which can be the destination address or a label, to find the corresponding output interface number in the forwarding table. Figure 18.2 shows the idea of the forwarding process in a router  Network Layer-I: Other Services Error Control Although error control also can be implemented in the network layer, the designers of the network layer in the Internet ignored this issue for the data being carried by the network layer. One reason for this decision is the fact that the packet in the network layer may be fragmented at each router, which makes error checking at this layer inefficient. Flow Control Flow control regulates the amount of data a source can send without overwhelming the receiver. If the upper layer at the source computer produces data faster than the upper layer at the destination computer can consume it, the receiver will be overwhelmed with data. To control the flow of data, the receiver needs to send some feedback to the sender to inform the latter that it is overwhelmed with data. The network layer in the Internet, however, does not directly provide any flow control. The datagrams are sent by the sender when they are ready, without any attention to the readiness of the receiver.  Network Layer-I: A few reasons for the lack of flow control in the design of the network layer can be mentioned. First, since there is no error control in this layer, the job of the network layer at the receiver is so simple that it may rarely be overwhelmed. Second, the upper layers that use the service of the network layer can implement buffers to receive data from the network layer as they are ready and do not have to consume the data as fast as it is received. Third, flow control is provided for most of the upper-layer protocols that use the services of the network layer, so another level of flow control makes the network layer more complicated and the whole system less efficient. Flow control is design issue at Data Link Layer. It is a technique that generally observes the proper flow of data from sender to receiver.  Network Layer-I: Congestion Control Another issue in a network-layer protocol is congestion control. Congestion in the network layer is a situation in which too many datagrams are present in an area of the Internet. Congestion may occur if the number of datagrams sent by source computers is beyond the capacity of the network or routers. In this situation, some routers may drop some of the datagrams. However, as more datagrams are dropped, the situation may become worse because, due to the error control mechanism at the upper layers, the sender may send duplicates of the lost packets. If the congestion continues, sometimes a situation may reach a point where the system collapses and no datagrams are delivered.  Network Layer-I: The reliance on individual routers to make routing decisions means each access point on the route must maintain a database of preferable directions for each ultimate destination. This disconnected strategy works most of the time. However, one router cannot know instantly if another router further down the line is overloaded or defective. All routers periodically inform their neighboring devices of status conditions. A problem at one point ripples through to recalculations performed in neighboring routers. Sometimes a router will calculate the best path and send a packet down a blocked route. By the time the packet approaches that block, the routers closer to the problem will already know about it and reroute the packet around the defective neighbor. That rerouting can overload alternative routers.  Network Layer-I: Quality of Service As the Internet has allowed new applications such as multimedia communication (in particular real-time communication of audio and video), the quality of service (QoS) of the communication has become more and more important. The Internet has thrived by providing better quality of service to support these applications. However, to keep the network layer untouched, these provisions are mostly implemented in the upper layer.  Network Layer-I: PROVIDING QoS IN THE INTERNET To support real-time audio and video communications the Internet must provide some level of end-to-end QoS. One approach provides differentiated service in the sense that some classes of traffic are treated preferentially relative to other classes. Packets are marked at the edge of the network to indicate the type of treatment that they are to receive in the routers inside the network. This approach does not provide strict QoS guarantees. A second approach provides guaranteed service that gives a strict bound on the end-to-end delay experienced by all packets that belong to a specific flow. This approach requires making resource reservations in the routers along the route followed by the given packet flow. Weighted fair queueing combined with traffic regulators are needed in the routers to provide this type of service.  Network Layer-I: Security Another issue related to communication at the network layer is security. Security was not a concern when the Internet was originally designed because it was used by a small number of users at universities for research activities; other people had no access to the Internet. The network layer was designed with no security provision. Today, however, security is a big concern. To provide security for a connectionless network layer, we need to have another virtual level that changes the connectionless service to a connection-oriented service. This virtual layer, called IPSec.  Network Layer-I: PACKET SWITCHING Switching: process of forwarding packet between devices on a N/W based on dest addr. Although in data communication switching techniques are divided into two broad categories, circuit switching and packet switching, only packet switching is used at the network layer because the unit of data at this layer is a packet. Circuit switching is mostly used at the physical layer; the electrical switch mentioned earlier is a kind of circuit switch. At the network layer, a message from the upper layer is divided into manageable packets and each packet is sent through the network. The source of the message sends the packets one by one; the destination of the message receives the packets one by one. The destination waits for all packets belonging to the same message to arrive before delivering the message to the upper layer.  Network Layer-I: The connecting devices in a packet-switched network still need to decide how to route the packets to the final destination. Today, a packet-switched network can use two different approaches to route the packets: the datagram approach and the virtual circuit approach. Datagram Approach: Connectionless Service When the Internet started, to make it simple, the network layer was designed to provide a connectionless service in which the network- layer protocol treats each packet independently, with each packet having no relationship to any other packet. The idea was that the network layer is only responsible for delivery of packets from the source to the destination. In this approach, the packets in a message may or may not travel the same path to their destination. Figure 18.3 shows the idea.  Network Layer-I: Each packet traveling in the Internet is an independent entity; there is no relationship between packets belonging to the same message. The switches in this type of network are called routers. A packet belonging to a message may be followed by a packet belonging to the same message or to a different message. Each packet is routed based on the information contained in its header: source and destination addresses. The destination address defines where it should go; the source address defines where it comes from. The router in this case routes the packet based only on the destination address. The source address may be used to send an error message to the source if the packet is discarded. Figure 18.4 shows the forwarding process in a router in this case. We have used symbolic addresses such as A and B.  Network Layer-I: Virtual-Circuit Approach: Connection-Oriented Service In a connection-oriented service (also called virtual-circuit approach), there is a relationship between all packets belonging to a message. Before all datagrams in a message can be sent, a virtual connection should be set up to define the path for the datagrams. After connection setup, the datagrams can all follow the same path. In this type of service, not only must the packet contain the source and destination addresses, it must also contain a flow label, a virtual circuit identifier that defines the virtual path the packet should follow. LABEL refers to a unique identifier associated with particular path or connection established between the source & destination. It is used to forward packets along the predefined path within a network. Use of LABEL makes source & destination addr unnecessary during data transfer, but part of the packet still using connectionless service and protocol at this layer is designed with these adrress.  Network Layer-I: In this case, the forwarding decision is based on the value of the label, or virtual circuit identifier, as it is sometimes called. To create a connection-oriented service, a three-phase process is used: setup, data transfer, and teardown. In the setup phase, the source and destination addresses of the sender and receiver are used to make table entries for the connection- oriented service. In the teardown phase, the source and destination inform the router to delete the corresponding entries. Data transfer occurs between these two phases.  Each packet is forwarded based on the label in the packet. To follow the idea of connection-oriented design to be used in the Internet, we assume that the packet has a label when it reaches the router. Figure 18.6 shows the idea. In this case, the forwarding decision is based on the value of the label, or virtual circuit identifier, as it is sometimes called. To create a connection-oriented service, a three-phase process is used: setup, data transfer, and teardown. In the setup phase, the source and destination addresses of the sender and receiver are used to make table entries for the connection-oriented service. In the teardown phase, the source and destination inform the router to delete the corresponding entries. Data transfer occurs between these two phases.  Network Layer-I: Setup Phase In the setup phase, a router creates an entry for a virtual circuit. For example, suppose source A needs to create a virtual circuit to destination B. Two auxiliary packets need to be exchanged between the sender and the receiver: the request packet and the acknowledgment packet. Request packet A request packet is sent from the source to the destination. This auxiliary packet carries the source and destination addresses. Figure 18.7 shows the process.  Network Layer-I: 1. Source A sends a request packet to router R1. 2. Router R1 receives the request packet. It knows that a packet going from A to B goes out through port 3. How the router has obtained this information is a point covered later. For the moment, assume that it knows the output port. The router creates an entry in its table for this virtual circuit, but it is only able to fill three of the four columns. The router assigns the incoming port (1) and chooses an available incoming label (14) and the outgoing port (3). It does not yet know the outgoing label, which will be found during the acknowledgment step. The router then forwards the packet through port 3 to router R3. 3. Router R3 receives the setup request packet. The same events happen here as at router R1; three columns of the table are completed: in this case, incoming port (1), incoming label (66), and outgoing port (3).  Network Layer-I: 4. Router R4 receives the setup request packet. Again, three columns are completed: incoming port (1), incoming label (22), and outgoing port (4). 5. Destination B receives the setup packet, and if it is ready to receive packets from A, it assigns a label to the incoming packets that come from A, in this case 77, as shown in Figure 18.8. This label lets the destination know that the packets come from A, and not from other sources. Acknowledgment Packet A special packet, called the acknowledgment packet, completes the entries in the switching tables. Figure 18.8 shows the process.  Network Layer-I: 1. The destination sends an acknowledgment to router R4. The acknowledgment carries the global source and destination addresses so the router knows which entry in the table is to be completed. The packet also carries label 77, chosen by the destination as the incoming label for packets from A. Router R4 uses this label to complete the outgoing label column for this entry. Note that 77 is the incoming label for destination B, but the outgoing label for router R4. 2. Router R4 sends an acknowledgment to router R3 that contains its incoming label in the table, chosen in the setup phase. Router R3 uses this as the outgoing label in the table. 3. Router R3 sends an acknowledgment to router R1 that contains its incoming label in the table, chosen in the setup phase. Router R1 uses this as the outgoing label in the table.  Network Layer-I: 4. Finally router R1 sends an acknowledgment to source A that contains its incoming label in the table, chosen in the setup phase. 5. The source uses this as the outgoing label for the data packets to be sent to destination B. Data-Transfer Phase The second phase is called the data-transfer phase. After all routers have created their forwarding table for a specific virtual circuit, then the network-layer packets belonging to one message can be sent one after another. In Figure 18.9, we show the flow of a single packet, but the process is the same for 1, 2, or 100 packets. The source computer uses the label 14, which it has received from router R1 in the setup  Network Layer-I: phase. Router R1 forwards the packet to router R3, but changes the label to 66. Router R3 forwards the packet to router R4, but changes the label to 22. Finally, router R4 delivers the packet to its final destination with the label 77. All the packets in the message follow the same sequence of labels, and the packets arrive in order at the destination. Teardown Phase In the teardown phase, source A, after sending all packets to B, sends a special packet called a teardown packet. Destination B responds with a confirmation packet. All routers delete the corresponding entries from their tables. Comparison Datagram Approach Virtual Circuit Approach connectionless Connection oriented Resources not reserved(on demand),if Resources reserved resources available it provided to packet Same route may or may not. Always takes Same route Only data transfer Set up, data transfer, tear down(delete path) Not much costly costly Not reliable reliable Forwarding based on dest addr. Forwarding based on flow label  Network Layer-I: NETWORK-LAYER PERFORMANCE The upper-layer protocols that use the service of the network layer expect to receive an ideal service, but the network layer is not perfect. The performance of a network can be measured in terms of few attributes i.e. delay, throughput, and packet loss. Congestion control is an issue that can improve the performance. Delay Delay should be minimum. Due to certain components in the N/W, Delay will be there to reach packet at receiver. All of us expect instantaneous response from a network, but a packet, from its source to its destination, encounters delays. The delays in a network can be divided into four types: transmission delay, propagation delay, processing delay, and queuing delay.  Network Layer-I: Transmission Delay: It is the time taken for packet to travel from sender until reaches to destination. A source host or a router cannot send a packet instantaneously. A sender needs to put the bits in a packet on the line one by one. Bigger packet means more time needed to place the packet on link. Sender at which rate transmitting packet on wire(medium) is the transmission rate. If the first bit of the packet is put on the line at time t1 and the last bit is put on the line at time t2, transmission delay of the packet is (t2 − t1). Definitely, the transmission delay is longer for a longer packet and shorter if the sender can transmit faster. In other words, the transmission delay is Delaytr = (Packet length) / (Transmission rate). Example: In a Fast Ethernet LAN with the transmission rate of 100 million bits per  Network Layer-I: = 10,000/ 100 x 106 = 0.0001 sec or 100 microseconds Propagation Delay It depends on the distance & propagation speed.(depend on medium) Propagation delay is the time it takes for a bit to travel from point A to point B in the transmission media. The propagation delay for a packet-switched network depends on the propagation delay of each network (LAN or WAN). The propagation delay depends on the propagation speed of the media, which is 3 × 108 meters/second in a vacuum and normally much less in a wired medium; it also depends on the distance of the link. In other words, propagation delay is Delaypg = (Distance) / (Propagation speed). Example 2 if the distance of a cable link in a point-to-point WAN is 2000 meters and the propagation speed of the bits in the cable is 2 × 10 8 meters/second, then the propagation delay is  Network Layer-I: Delaypg = (Distance) / (Propagation speed). = 2000 / 2 x 108 = 0.000001 sec or 10 microseconds Processing Delay Each component in N/W has to process i.e. packet starts from sender& move towards receiver in between other devices like router. Each router has input port & output port. The processing delay is the time required for a router or a destination host to receive a packet from its input port, remove the header, perform an error detection procedure, and deliver the packet to the output port (in the case of a router) or deliver the packet to the upper-layer protocol (in the case of the destination host). The processing delay may be different for each packet, but normally is calculated as an average. Delaypr = Time required to process a packet in a router or a destination host  Network Layer-I: Queuing Delay Queuing delay can normally happen in a router. As we discuss in the next section, a router has an input queue connected to each of its input ports to store packets waiting to be processed; the router also has an output queue connected to each of its output ports to store packets waiting to be transmitted. The queuing delay for a packet in a router is measured as the time a packet waits in the input queue and output queue of a router. We can compare the situation with a busy airport. Some planes may need to wait to get the landing band (input delay); some planes may need to wait to get the departure band (output delay). Delayqu = The time a packet waits in input and output queues in a router  Network Layer-I: Total Delay Assuming equal delays for the sender, routers, and receiver, the total delay (source-to destination delay) a packet encounters can be calculated if we know the number of routers, n, in the whole path. Note that if we have n routers, we have (n + 1) links. Therefore, we have (n + 1) transmission delays related to n routers and the source, (n + 1) propagation delays related to (n + 1) links, (n + 1) processing delays related to n routers and the destination, and only n queuing delays related to n routers. Q: If there are n routers why do we need n+1 links  Network Layer-I: If there are n routers in a network, you need n+1 links to connect them all in a way that allows all routers to communicate with each other. This is because each router needs to be connected to at least one other router to be part of the network, and a minimum of two links are required to connect two routers. Router 1 --- Router 2 --- Router 3 To connect all three routers, we need at least one additional link. One possible way to achieve this is to add a link between Router 1 and Router 3, resulting in a triangular network with three links: Router 2 / \ / \ Router 1 --- Router 3  Network Layer-I: Throughput Throughput at any point in a network is defined as the number of bits passing through the point in a second, which is actually the transmission rate of data at that point. In a path from source to destination, a packet may pass through several links (networks), each with a different transmission rate. Throughput: how many bits per second are going through the network How, then, can we determine the throughput of the whole path? To see the situation, assume that we have three links, each with a different transmission rate, as shown in Figure 18.10. In this figure, the data can flow at the rate of 200 kbps in Link1. However, when the data arrives at router R1, it cannot pass at this rate. Data needs to be queued at the router and sent at 100 kbps. When data arrives at router R2, it could be sent at the rate  Network Layer-I: of 150 kbps, but there is not enough data to be sent. In other words, the average rate of the data flow in Link3 is also 100 kbps. We can conclude that the average data rate for this path is 100 kbps, the minimum of the three different data rates. In general, in a path with n links in series, we have Packet Loss Another issue that severely affects the performance of communication is the number of packets lost during transmission. When a router receives a packet while processing another packet, the received packet needs to be stored in the input buffer waiting for its turn. A router, however, has an input buffer with a limited size.  Network Layer-I: A time may come when the buffer is full and the next packet needs to be dropped. The effect of packet loss on the Internet network layer is that the packet needs to be resent, which in turn may create overflow and cause more packet loss. Congestion Control Congestion control is a mechanism for improving performance. Although congestion at the network layer is not explicitly addressed in the Internet model, the study of congestion at this layer may help us to better understand the cause of congestion at the transport layer and find possible remedies to be used at the network layer. Congestion at the network layer is related to two issues, throughput and delay,  Network Layer-I: When the load is much less than the capacity of the network, the delay is at a minimum. This minimum delay is composed of propagation delay and processing delay, both of which are negligible. However, when the load reaches the network capacity, the delay increases sharply because we now need to add the queuing delay to the total delay. Note that the delay becomes infinite when the load is greater than the capacity. When the load is below the capacity of the network, the throughput increases proportionally with the load. We expect the throughput to remain constant after the load reaches the capacity, but instead the throughput declines sharply. The reason is the discarding of packets by the routers. When the load exceeds the capacity, the queues become full and the routers have to discard some packets.  Network Layer-I: Congestion Control Congestion control refers to techniques and mechanisms that can either prevent congestion before it happens or remove congestion after it has happened. In general, we can divide congestion control mechanisms into two broad categories: open-loop congestion control (prevention) and closed-loop congestion control (removal). Open-Loop Congestion Control In open-loop congestion control, policies are applied to prevent congestion before it happens. In these mechanisms, congestion control is handled by either the source or the destination. We give a brief list of policies that can prevent congestion. Retransmission Policy Retransmission is sometimes unavoidable. If the sender feels that a sent packet is lost or corrupted, the packet needs to be retransmitted. Retransmission in general may increase congestion in the network.  Network Layer-I: However, a good retransmission policy can prevent congestion. The retransmission policy and the retransmission timers must be designed to optimize efficiency and at the same time prevent congestion. Window Policy The type of window at the sender may also affect congestion. The Selective Repeat window is better than the Go-Back-N window for congestion control. In the Go-Back-N window, when the timer for a packet times out, several packets may be resent, although some may have arrived safe and sound at the receiver. This duplication may make the congestion worse. The Selective Repeat window, on the other hand, tries to send the specific packets that have been lost or corrupted. Acknowledgment Policy The acknowledgment policy imposed by the receiver may also affect congestion  Network Layer-I:. If the receiver does not acknowledge every packet it receives, it may slow down the sender and help prevent congestion. Several approaches are used in this case. A receiver may send an acknowledgment only if it has a packet to be sent or a special timer expires. A receiver may decide to acknowledge only N packets at a time. Discarding Policy A good discarding policy by the routers may prevent congestion and at the same time may not harm the integrity of the transmission. For example, in audio transmission, if the policy is to discard less sensitive packets when congestion is likely to happen, the quality of sound is still preserved and congestion is prevented or alleviated.  Network Layer-I: Admission Policy An admission policy, which is a quality-of-service mechanism (discussed in Chapter 30), can also prevent congestion in virtual-circuit networks. Switches in a flow first check the resource requirement of a flow before admitting it to the network. A router can deny establishing a virtual-circuit connection if there is congestion in the network or if there is a possibility of future congestion. Closed-Loop Congestion Control Closed-loop congestion control mechanisms try to alleviate congestion after it happens. Backpressure The technique of backpressure refers to a congestion control mechanism in which a congested node stops receiving data from the immediate upstream node or nodes. This may cause the upstream node or nodes to become congested, and they, in turn, reject data from their upstream node or nodes, and so on.  Network Layer-I: Backpressure is a node to-node congestion control that starts with a node and propagates, in the opposite direction of data flow, to the source. The backpressure technique can be applied only to virtual circuit networks, in which each node knows the upstream node from which a flow of data is coming.  Network Layer-I: Node III in the figure has more input data than it can handle. It drops some packets in its input buffer and informs node II to slow down. Node II, in turn, may be congested because it is slowing down the output flow of data. If node II is congested, it informs node I to slow down, which in turn may create congestion. If so, node I informs the source of data to slow down. This, in time, alleviates the congestion. Note that the pressure on node III is moved backward to the source to remove the congestion. Choke Packet A choke packet is a packet sent by a node to the source to inform it of congestion.  Network Layer-I: In the choke-packet method, the warning is from the router, which has encountered congestion, directly to the source station. The intermediate nodes through which the packet has traveled are not warned. An example of this type of control in ICMP (discussed in Chapter 19). When a router in the Internet is overwhelmed with IP datagrams, it may discard some of them, but it informs the source host, using a source quench ICMP message. The warning message goes directly to the source station; the intermediate routers do not take any action. Figure 18.15 shows the idea of a choke packet.  Network Layer-I: Implicit Signaling In implicit signaling, there is no communication between the congested node or nodes and the source. The source guesses that there is congestion somewhere in the network from other symptoms. For example, when a source sends several packets and there is no acknowledgment for a while, one assumption is that the network is congested. The delay in receiving an acknowledgment is interpreted as congestion in the network; the source should slow down. Explicit Signaling The node that experiences congestion can explicitly send a signal to the source or destination. The explicit-signaling method, however, is different from the choke-packet method. In the choke-packet method, a separate packet is used for this purpose; in the explicit-signaling method, the signal is included in the packets that carry data. Explicit signaling can occur in either the forward or the backward direction. This type of congestion control can be seen in an ATM network,  Network Layer-I: Explicit Congestion Notification (ECN) ECN field: A 2-bit field specified for use with explicit congestion signaling in the IPv4 and IPv6 packet headers. IPV4 ADDRESSES An IPv4 address is a 32-bit address that uniquely and universally defines the connection of a host or a router to the Internet. IPv4 addresses are unique in the sense that each address defines one, and only one, connection to the Internet. If a device has two connections to the Internet, via two networks, it has two IPv4 addresses. IPv4 addresses are universal in the sense that the addressing system must be accepted by any host that wants to be connected to the Internet. Address Space A protocol like IPv4 that defines addresses has an address space. An address space is the total number of addresses used by the protocol.  Network Layer-I: If a protocol uses b bits to define an address, the address space is 2b because each bit can have two different values (0 or 1). IPv4 uses 32-bit addresses, which means that the address space is 232 or 4,294,967,296 (more than four billion). Notation There are three common notations to show an IPv4 address: binary notation (base 2), dotted-decimal notation (base 256), and hexadecimal notation (base 16). In binary notation, an IPv4 address is displayed as 32 bits. To make the address more readable, one or more spaces are usually inserted between each octet (8 bits). Each octet is often referred to as a byte. To make the IPv4 address more compact and easier to read, it is usually written in decimal form with a decimal point (dot) separating the bytes. This format is referred to as dotted-decimal notation.  Network Layer-I: Note that because each byte (octet) is only 8 bits, each number in the dotted-decimal notation is between 0 and 255. We sometimes see an IPv4 address in hexadecimal notation. Each hexadecimal digit is equivalent to four bits. This means that a 32-bit address has 8 hexadecimal digits. This notation is often used in network programming.  Network Layer-I: Hierarchy in Addressing In any communication network that involves delivery, such as a telephone network or a postal network, the addressing system is hierarchical. In a postal network, the postal address (mailing address) includes the country, state, city, street, house number, and the name of the mail recipient. Similarly, a telephone number is divided into the country code, area code, local exchange, and the connection. A 32-bit IPv4 address is also hierarchical, but divided only into two parts. The first part of the address, called the prefix, defines the network; the second part of the address, called the suffix, defines the node (connection of a device to the Internet). Figure 18.17 shows the prefix and suffix of a 32-bit IPv4 address. The prefix length is n bits and the suffix length is (32 − n) bits.  Network Layer-I: A prefix can be fixed length or variable length. The network identifier in the IPv4 was first designed as a fixed-length prefix. This scheme, which is now obsolete, is referred to as classful addressing. The new scheme, which is referred to as classless addressing, uses a variable-length network prefix.  Network Layer-I: Classful Addressing When the Internet started, an IPv4 address was designed with a fixed- length prefix, but to accommodate both small and large networks, three fixed-length prefixes were designed instead of one (n = 8, n = 16, and n = 24). The whole address space was divided into five classes (class A, B, C, D, and E), as shown in Figure 18.18. This scheme is referred to as classful addressing. In class A, the network length is 8 bits, but since the first bit, which is 0, defines the class, we can have only seven bits as the network identifier. This means there are only 27 = 128 networks in the world that can have a class A address. In class B, the network length is 16 bits, but since the first two bits, which are (10)2, define the class, we can have only 14 bits as the network identifier. This means there are only 214 = 16,384 networks in the world that can have a class B address.  Network Layer-I: All addresses that start with (110)2 belong to class C. In class C, the network length is 24 bits, but since three bits define the class, we can have only 21 bits as the network identifier. This means there are 221 = 2,097,152 networks in the world that can have a class C address. Class D is used for multicast addresses. All addresses that start with 1111 in binary belong to class E. Address Depletion The reason that classful addressing has become obsolete is address depletion. To understand the problem, let us think about class A. This class can be assigned to only 128 organizations in the world, but each organization needs to have a single network (seen by the rest of the world) with 16,777,216 nodes (computers in this single network). Since there may be only a few organizations that are this large, most of the addresses in this class were wasted (unused). Class B addresses were designed for midsize organizations, but many of the addresses in this class also remained unused.  Network Layer-I: Class C addresses have a completely different flaw in design. The number of addresses that can be used in each network (256) was so small that most companies were not comfortable using a block in this address class. Class E addresses were almost never used, wasting the whole class. Subnetting and Supernetting To alleviate address depletion, two strategies were proposed and, to some extent, implemented: subnetting and supernetting. In subnetting, a class A or class B block is divided into several subnets. Each subnet has a larger prefix length than the original network. Supernetting was devised to combine several class C blocks into a larger block to be attractive to organizations that need more than the 256 addresses available in a class C block.  Network Layer-I: Advantage of Classful Addressing Although classful addressing had several problems and became obsolete, it had one advantage: Given an address, we can easily find the class of the address and, since the prefix length for each class is fixed, we can find the prefix length immediately. Classless Addressing Subnetting and supernetting in classful addressing did not really solve the address depletion problem. With the growth of the Internet, it was clear that a larger address space was needed as a long-term solution. The larger address space, however, requires that the length of IP addresses also be increased, which means the format of the IP packets needs to be changed. Although the long-range solution has already been devised and is called IPv6 (discussed later), a short-term solution was also devised to use the same address space but to change the distribution of addresses to provide a fair share to each organization.  Network Layer-I: The short-term solution still uses IPv4 addresses, but it is called classless addressing. In other words, the class privilege was removed from the distribution to compensate for the address depletion. Because of fixed prefix length, got address depletion problem, so we don’t have enough addresses to allocate IP addresses to all system. In 1996, the Internet authorities announced a new architecture called classless addressing. In classless addressing, variable-length blocks are used that belong to no classes. Representation of classless addresses: We can have a block of 1 address, 2 addresses, 4 addresses, 128 addresses, and so on. One of the restrictions, as we discuss later, is that the number of addresses in a block needs to be a power of 2. An organization can be granted one block of addresses. Unlike classful addressing, the prefix length in classless addressing is variable. We can have a prefix length that ranges from 0 to 32. The size of the network is inversely proportional to the length of the prefix. A small prefix means a larger network; a large prefix means a smaller network.  Network Layer-I: We need to emphasize that the idea of classless addressing can be easily applied to classful addressing. How to represent classless addresses? Prefix Length: Slash Notation The first question that we need to answer in classless addressing is how to find the prefix length if an address is given.. In this case, the prefix length, n, is added to the address, separated by a slash. The notation is informally referred to as slash notation and formally as classless interdomain routing or CIDR (pronounced cider) strategy. An address in classless addressing can then be represented as shown in Figure 18.20.  Network Layer-I: Extracting Information from an Address Given any address in the block, we normally like to know three pieces of information about the block to which the address belongs: the number of addresses, the first address in the block, and the last address. Since the value of prefix length, n, is given, we can easily find these three pieces of information, as shown in Figure 18.21. 1. The number of addresses in the block is found as N = 232−n. (number of IP address in host) 2. To find the first address, we keep the n leftmost bits and set the (32 − n) rightmost bits all to 0s. 3. To find the last address, we keep the n leftmost bits and set the (32 − n) rightmost bits all to 1s.  Network Layer-I: Example 18.1 A classless address is given as 167.199.170.82/27. We can find the above three pieces of information as follows. The number of addresses in the network is 232 − n = 25 = 32 addresses. (32 addresses can be allocated) Network Layer-I:  Network Layer-I: Network Address The first address, the network address, is particularly important because it is used in routing a packet to its destination network.  Identify the address class of the following IP addresses: 200.58.20.165;128.167.23.20; 16.196.128.50;150.156.10.10;250.10.24.96. 200.58.20.165 11001000.00111010.00010100.10100101 Class C 128.167.23.20 10000000.10100111.00010111.00010100 Class B 16.196.128.50 00010000.11000100.10000000.00110010 Class A 150.156.10.10 10010110.10011100.00001010.00001010 Class B 250.10.24.96 11111010.00001010 ? Network Layer-I: Network Layer-I:  Network Layer-I: Here ISP granted block of address 14.24.74.0/24 & organization need to divide this into 3 subblock. DHCP Mainly useful in order to assign IP addr to host.). If u use internet &u wont use DHCP protocol then N/w admin hat to enter all the details. Assume organization has 10000 hosts then then internet people has to enter all above details in all 10000 hosts, which is time consuming. To overcome this problem,we use DHCP protocol. DHCP server contains pools of IP addr. Duty of server to assign IP addr to all hosts in an organization. DHCP server provides all details(IP addr, subnet mask,default gateway, preferred DNS server, alternate dns server. )  Network Layer-I: Dynamic Host Configuration Protocol (DHCP) However, address assignment in an organization can be done automatically using the Dynamic Host Configuration Protocol (DHCP). Usually large organization receive blockof IP addr(want to connect to internet) from ICANN. (provide IP addr to any organization.) If small organization means receive address from ISP. After block of of IP addr assigned, n/w admin has to ,manually assign IP adr. To individual host. What DHCP do? If organization has DHCP protocol.addr assignment done automatiacally without n/w admin. DHCP is an application-layer program, using the client-server paradigm, that actually helps TCP/IP at the network layer. DHCP has found such widespread use in the Internet that it is often called a plug and- play protocol. In can be used in many situations. A network manager can configure DHCP to assign permanent IP addresses to the host and routers. DHCP can also be configured to provide temporary, on demand, IP addresses to hosts.  Operations on DHCP Whenever C wants IP addr from any of server,it uses DHCP protocol.i.e. C doesnot know his IP addr. First C sends Discover msg to all nearby server(broadcast msg) for IP addr. If any of server interested to gv their own IP addr they give DHCP offer msg to C. Now C get many offer,among those it select one, and send DHCP REQUEST to specified server. DHCP server send ACK to C if they want to gv their IP addr So whenever CLIENT want IP addr from SERVER,it sends DHCPDISCOVER msg to SERVER (BROADCASTE MESSAGE) SERVER sends best OFFER with IP addr. CLIENT select any one of the offer & REQUEST the SERVER. SEREVR send ACK MSG to CLIENT by accepting their REQUEST. CLIENT use IP Addr. For specified time  Network Layer-I: 1. The joining host creates a DHCPDISCOVER message in which only the transaction- ID field is set to a random number. No other field can be set because the host has no knowledge with which to do so. This message is encapsulated in a UDP user datagram with the source port set to 68 and the destination port set to 67. We will discuss the reason for using two well-known port numbers later. The user datagram is encapsulated in an IP datagram with the source address set to 0.0.0.0 (“this host”) and the destination address set to 255.255.255.255 (broadcast address). The reason is that the joining host knows neither its own address nor the server address. 2. The DHCP server or servers (if more than one) responds with a DHCPOFFER message in which the your address field defines the offered IP address for the joining host and the server address field includes the IP address of the server. The message also includes the lease time for which the host can keep the IP address.  Network Layer-I: This message is encapsulated in a user datagram with the same port numbers, but in the reverse order. The user datagram in turn is encapsulated in a datagram with the server address as the source IP address, but the destination address is a broadcast address, in which the server allows other DHCP servers to receive the offer and give a better offer if they can. 3. The joining host receives one or more offers and selects the best of them. The joining host then sends a DHCPREQUEST message to the server that has given the best offer. The fields with known value are set. The message is encapsulated in a user datagram with port numbers as the first message. The user datagram is encapsulated in an IP datagram with the source address set to the new client address, but the destination address still is set to the broadcast address to let the other servers know that their offer was not accepted.  Network Layer-I: 4. Finally, the selected server responds with a DHCPACK message to the client if the offered IP address is valid. If the server cannot keep its offer (for example, if the address is offered to another host in between), the server sends a DHCPNACK message and the client needs to repeat the process. This message is also broadcast to let other servers know that the request is accepted or rejected.  Network Layer-I: Forwarding IP Packet : based on Destination Address: PLEASE REFER TEXTBOOK  Network Layer-I: INTERNET PROTOCOL (IP) The network layer in version 4 can be thought of as one main protocol and three auxiliary ones. The main protocol, Internet Protocol version 4 (IPv4), is responsible for packetizing, forwarding, and delivery of a packet at the network layer. The Internet Control Message Protocol version 4 (ICMPv4) helps IPv4 to handle some errors that may occur in the network-layer delivery. The Internet Group Management Protocol (IGMP) is used to help IPv4 in multicasting. The Address Resolution Protocol (ARP) is used to glue the network and data-link layers in mapping network-layer addresses to link-layer addresses. Figure 19.1 shows the positions of these four protocols in the TCP/IP protocol suite  Network Layer-I: IPv4 is an unreliable datagram protocol—a best-effort delivery service. The term best-effort means that IPv4 packets can be corrupted, be lost, arrive out of order, or be delayed, and may create congestion for the network. If reliability is important, IPv4 must be paired with a reliable transport- layer protocol such as TCP. An example of a more commonly understood best-effort delivery service is the post office. The post office does its best to deliver the regular mail but does not always succeed. If an unregistered letter is lost or damaged, it is up to the sender or would-be recipient to discover this. The post office itself does not keep track of every letter and cannot notify a sender of loss or damage of one.  Network Layer-I: IPv4 is also a connectionless protocol that uses the datagram approach. This means that each datagram is handled independently, and each datagram can follow a different route to the destination. This implies that datagrams sent by the same source to the same destination could arrive out of order. Again, IPv4 relies on a higher-level protocol to take care of all these problems. Datagram Format Packets used by the IP are called datagrams. Figure 19.2 shows the IPv4 datagram format. A datagram is a variable-length packet consisting of two parts: header and payload (data). The header is 20 to 60 bytes in length and contains information essential to routing and delivery. It is customary in TCP/IP to show the header in 4-byte sections.  Network Layer-I: Version: The version field indicates the version number used by the IP packet so that revisions can be distinguished from each other. The current IP version is 4. Version 5 is used for a real-time stream protocol called ST2, and version 6 is used for the new generation IP known as IPv6. Header Length. The 4-bit header length (HLEN) field defines the total length of the datagram header in 4-byte words. The IPv4 datagram has a variable-length header. When a device receives a datagram, it needs to know when the header stops and the data, which is encapsulated in the packet, starts. However, to make the value of the header length (number of bytes) fit in a 4- bit header length, the total length of the header is calculated as 4-byte words. The total length is divided by 4 and the value is inserted in the field. The receiver needs to multiply the value of this field by 4 to find the total length.  Network Layer-I: Type of service: The type of service (TOS) field traditionally specifies the priority of the packet based on delay, throughput, reliability, and cost requirements. Three bits are assigned for priority levels (called “precedence”) and four bits for the specific requirement (i.e., delay, throughput, reliability, and cost). For example, if a packet needs to be delivered to the destination as soon as possible, the transmitting IP module can set the delay bit to one and use a high- priority level. The TOS field is not in common use and so the field is usually set to zero. Recent work in the Differentiated Services Working Group of IETF redefines the TOS field in order to support other services that are better than the basic best effort. Total Length. This 16-bit field defines the total length (header plus data) of the IP datagram in bytes. A 16-bit number can define a total length of up to 65,535 (when all bits are 1s). However, the size of the datagram is normally much less than this. This field helps the receiving device to know when the packet has completely arrived.  Network Layer-I: To find the length of the data coming from the upper layer, subtract the header length from the total length. The header length can be found by multiplying the value in the HLEN field by 4. In practice the maximum possible length is very rarely used, since most physical networks have their own length limitation. For example, Ethernet limits the payload length to 1500 byte Identification, Flags, and Fragmentation Offset. These three fields are related to the fragmentation of the IP datagram when the size of the datagram is larger than the underlying network can carry.  Network Layer-I: Time-to-live. The time-to-live (TTL) field is used to control the maximum number of hops (routers) visited by the datagram. When a source host sends the datagram, it stores a number in this field. This value is approximately two times the maximum number of routers between any two hosts. Each router that processes the datagram decrements this number by one. If this value, after being decremented, is zero, the router discards the datagram. Protocol: The protocol field specifies the upper-layer protocol that is to receive the IP data at the destination host. Examples of the protocols include TCP (protocol = 6), UDP (protocol = 17), and ICMP (protocol = 1).  Network Layer-I: Header checksum: The header checksum field verifies the integrity of the header of the IP packet. The data part is not verified and is left to upper-layer protocols. If the verification process fails, the packet is simply discarded. Note that when a router decrements the TTL field, the router must also recompute the header checksum field. Source IP address and destination IP address: These fields contain the addresses of the source and destination hosts. Options: The options field, which is of variable length, allows the packet to request special features such as security level, route to be taken by the packet, and timestamp at each router. For example, a source host can use the options field to specify a sequence of routers that a datagram is to traverse on its way to the destination host. Padding: This field is used to make the header a multiple of 32-bit words.  Network Layer-I: 1. An IPv4 packet has arrived with the first 8 bits as (01000010)2 The receiver discards the packet. Why? 2. In an IPv4 packet, the value of HLEN is (1000)2. How many bytes of options are being carried by this packet? 3. In an IPv4 packet, the value of HLEN is 5, and the value of the total length field is (0028)16. How many bytes of data are being carried by this packet?  Network Layer-I: Solution There is an error in this packet. The 4 leftmost bits (0100) 2 show the version, which is correct. The next 4 bits (0010)2 show an invalid header length (2 × 4 = 8). The minimum number of bytes in the header must be 20. The packet has been corrupted in transmission. Solution The HLEN value is 8, which means the total number of bytes in the header is 8 × 4, or 32 bytes. The first 20 bytes are the base header, the next 12 bytes are the options. Solution The HLEN value is 5, which means the total number of bytes in the header is 5 × 4, or 20 bytes (no options). The total length is (0028)16 or 40 bytes, which means the packet is carrying 20 bytes of data (40 − 20).  Network Layer-I: Fragmentation and Reassembly One of the strengths of IP is that it can work on a variety of physical networks. Each physical network usually imposes a certain packet-size limitation on the packets that can be carried, called the maximum transmission unit (MTU). For example, Ethernet specifies an MTU of 1500 bytes, and FDDI specifies an MTU of 4464 bytes.  Network Layer-I: When IP has to send a packet that is larger than the MTU of the physical network, IP must break the packet into smaller fragments whose size can be no larger than the MTU. Each fragment is sent independently to the destination as though it were an IP packet. If the MTU of some other network downstream is found to be smaller than the fragment size, the fragment will be broken again into smaller fragments, as shown in Figure 8.11.  Network Layer-I: The destination IP is the only entity that is responsible for reassembling the fragments into the original packet. To reassemble the fragments, the destination waits until it has received all the fragments belonging to the same packet. If one or more fragments are lost in the network, the destination abandons the reassembly process and discards the rest of the fragments. To detect lost fragments, the destination host sets a timer once the first fragment of a packet arrives. If the timer expires before all fragments have been received, the host assumes the missing fragments were lost in the network and discards the other fragments.  Network Layer Protocols-II ICMPv4 The IPv4 has no error-reporting or error-correcting mechanism. What happens if something goes wrong? What happens if a router must discard a datagram because it cannot find a route to the final destination, or because the time-to-live field has a zero value? What happens if the final destination host must discard the received fragments of a datagram because it has not received all fragments within a predetermined time limit? These are examples of situations where an error has occurred and the IP protocol has no built-in mechanism to notify the original host. The IP protocol also lacks a mechanism for host and management queries.  Function of IPV4: send packet from source to destination i.e. select a route to network, forward packet so it reaches to destination. IP not reliable. i.e. if any error during transmission,not reported to source host. Types of error during transmission? Router may drop packet if dest receives one fragment & fragment not received. TTL=0 means not intended receiver. All these errors must report to source host  Network Layer Protocols-II The Internet Control Message Protocol version 4 (ICMPv4) has been designed to compensate for the above two deficiencies. It is a companion to the IP protocol. ICMP itself is a network-layer protocol. However, its messages are not passed directly to the data-link layer as would be expected. Instead, the messages are first encapsulated inside IP datagrams before going to the lower layer. When an IP datagram encapsulates an ICMP message, the value of the protocol field in the IP datagram is set to 1 to indicate that the IP payroll is an ICMP message. MESSAGES ICMP messages are divided into two broad categories: error-reporting messages and query messages.  Network Layer Protocols-II The error-reporting messages report problems that a router or a host (destination) may encounter when it processes an IP packet. The query messages, which occur in pairs, help a host or a network manager get specific information from a router or another host. An ICMP message has an 8-byte header and a variable-size data section. Although the general format of the header is different for each message type, the first 4 bytes are common to all. The first field, ICMP type, defines the type of the message. The code field specifies the reason for the particular message type. The last common field is the checksum field. The rest of the header is specific for each message type.  Network Layer Protocols-II Error Reporting Messages Since IP is an unreliable protocol, one of the main responsibilities of ICMP is to report some errors that may occur during the processing of the IP datagram. ICMP does not correct errors, it simply reports them. Error correction is left to the higher-level protocols. Error messages are always sent to the original source because the only information available in the datagram about the route is the source and destination IP addresses. ICMP uses the source IP address to send the error message to the source (originator) of the datagram. To make the error-reporting process simple, ICMP follows some rules in reporting messages.  Network Layer Protocols-II First, no error message will be generated for a datagram having a multicast address or special address (such as this host or loopback). Second, no ICMP error message will be generated in response to a datagram carrying an ICMP error message. Third, no ICMP error message will be generated for a fragmented datagram that is not the first fragment. Example Destination Unreachable The most widely used error message is the destination unreachable (type 3). This message uses different codes (0 to 15) to define the type of error message and the reason why a datagram has not reached its final destination. For example, code 0 tells the source that a host is unreachable. This may happen, for example, when we use the HTTP protocol to access a web page, but the server is down. The message “destination host is not reachable” is created and sent back to the source.  Network Layer Protocols-II Source Quench Another error message is called the source quench (type 4) message, which informs the sender that the network has encountered congestion and the datagram has beenbdropped; the source needs to slow down sending more datagrams. In other words, ICMP adds a kind of congestion control mechanism to the IP protocol by using this type of message. Query Messages Interestingly, query messages in ICMP can be used independently without relation to an IP datagram. Of course, a query message needs to be encapsulated in a datagram, as a carrier. Query messages are used to probe or test the liveliness of hosts or routers in the Internet, find the one-way or the round-trip time for an IP datagram between two devices, or even find out whether the clocks in two devices are synchronized.  Network Layer Protocols-II Naturally, query messages come in pairs: request and reply. The echo request (type 8) and the echo reply (type 0) pair of messages are used by a host or a router to test the liveliness of another host or router. A host or router sends an echo request message to another host or router; if the latter is alive, it responds with an echo reply message. We shortly see the applications of this pair in two debugging tools: ping and traceroute. Ping We can use the ping program to find if a host is alive and responding. We use ping here to see how it uses ICMP packets. The source host sends ICMP echo-request messages; the destination, if alive, responds with ICMP echo-reply messages. The ping program sets the identifier field in the echo-request and echo-reply message and starts the sequence number from 0; this number is incremented by 1 each time a new message is sent.  Network Layer Protocols-II Note that ping can calculate the round-trip time. It inserts the sending time in the data section of the message. When the packet arrives, it subtracts the arrival time from the departure time to get the round-trip time (RTT). Traceroute or Tracert The traceroute program in UNIX or tracert in Windows can be used to trace the path of a packet from a source to the destination. It can find the IP addresses of all the routers that are visited along the path. The program is usually set to check for the maximum of 30 hops (routers) to be visited. The number of hops in the Internet is normally less than this.  Network Layer Protocols-II The traceroute program gets help from two error-reporting messages: time-exceeded and destination-unreachable. The traceroute is an application layer program, but only the client program is needed, because, as we can see, the client program never reaches the application layer in the destination host. In other words, there is no traceroute server program. The traceroute application program is encapsulated in a UDP user datagram, but traceroute intentionally uses a port number that is not available at the destination. If there are n routers in the path, the traceroute program sends (n + 1) messages. The first n messages are discarded by the n routers, one by each router; the last message is discarded by the destination host. The traceroute client program uses the (n + 1) ICMP error-reporting messages received to find the path between the routers.  Network Layer Protocols-II The first traceroute message is sent with time-to-live (TTL) value set to 1; the message is discarded at the first router and a time-exceeded ICMP error message is sent, from which the traceroute program can find the IP address of the first router (the source IP address of the error message) and the router name (in the data section of the message). The second traceroute message is sent with TTL set to 2, which canfind the IP address and the name of the second router. Similarly, the third message can find the information about router 3. The fourth message, however, reaches the destination host. This host is also dropped, but for another reason. The destination host cannot find the port number specified in the UDP user datagram.  Network Layer Protocols-II This time ICMP sends a different message, the destination-unreachable message with code 3 to show the port number is not found. After receiving this different ICMP message, the traceroute program knows that the final destination is reached. It uses the information in the received message to find the IP address and the name of the final destination. Slow hops range from 250ms to 300ms. If you receive asterisks, your results did not return within the TTL (time to live) value. This is due to the ICMP (Internet Control Message Protocol) blocking traffic or the packet did not reach the intended destination and timed out.