Networking Final Exam

Ethernet technologies provide an unreliable service to the network layer. Specifically, when adapter B receives a frame from adapter A, it runs the frame through a CRC check, but neither sends an acknowledgment when a frame passes the CRC check nor sends a negative acknowledgment when a frame fails the CRC check. When a frame fails the CRC check, adapter B simply discards the frame. Thus, adapter A has no idea whether its transmitted frame reached adapter B and passed the CRC check. This lack of reliable transport (at the link layer) helps to make Ethernet simple and cheap. But it also means that the stream of datagrams passed to the network layer can have gaps. If there are gaps due to discarded Ethernet frames, does the application at Host B see gaps as well? As we learned in Chapter 3, this depends on whether the application is using UDP or TCP. If the application is using UDP, then the application in Host B will indeed see gaps in the data. On the other hand, if the application is using TCP, then TCP in Host B will not acknowledge the data contained in discarded frames, causing TCP in Host A to retransmit. Note that when TCP retransmits data, the data will eventually return to the Ethernet adapter at which it was discarded. Thus, in this sense, Ethernet does retransmit data, although Ethernet is unaware of whether it is transmitting a brand-new datagram with brand-new data, or a datagram that contains data that has already been transmitted at least once.

Briefly describe ethernet technologies

Let's illustrate some of the basic concepts of BGP routing policy with a simple example. Figure 4.42 shows six interconnected autonomous systems: A, B, C, W, X, and Y. It is important to note that A, B, C, W, X, and Y are ASs, not routers. Let's assume that autonomous systems W, X, and Yare stub networks and that A, B, and C are backbone provider networks. We'll also assume that A, B, and C, all peer with each other, and provide full BGP information to their customer networks. All traffic entering a stub network must be destined for that network, and all traffic leaving a stub network must have originated in that network. W and Y are clearly stub networks. X is a multihomed stub network, since it is connected to the rest of the network via two different providers (a scenario that is becoming increasingly common in practice). However, like Wand Y, X itself must be the source/destination of all traffic leaving/entering X. But how will this stub network behavior be implemented and enforced? How will X be prevented from forwarding traffic between B and C? This can easily be accomplished by controlling the manner in which BGP routes are advertised. In particular, X will function as a stub network if it advertises (to its neighbors B and C) that it has no paths to any other destinations except itself. That is, even though X may know of a path, say XCY, that reaches network Y, it will not advertise this path to B. Since B is unaware that X has a path to Y, B would never forward traffic destined to Y (or C) via X. This simple example illustrates how a selective route advertisement policy can be used to implement customer/provider routing relationships (see page 397 for diagram)

Describe how stub, multi-homed stub networks, and provider AS's are related

http://www.dcs.bbk.ac.uk/~ptw/teaching/IWT/network-layer/datagram-format.gif █ Version number. These 4 bits specify the IP protocol version of the datagram. By looking at the version number, the router can determine how to interpret the remainder of the IP datagram. Different versions of IP use different datagram formats. The datagram format for the current version of IP, IPv4, is shown in Figure 4.13. The datagram format for the new version of IP (IPv6) is discussed at the end of this section. █ Header length. Because an IPv4 datagram can contain a variable number of options (which are included in the IPv4 datagram header), these 4 bits are needed to determine where in the IP datagram the data actually begins. Most IP datagrams do not contain options, so the typical IP datagram has a 20-byte header. █ Type of service. The type of service (TOS) bits were included in the IPv4 header to allow different types of IP datagrams (for example, datagrams particularly requiring low delay, high throughput, or reliability) to be distinguished from each other. For example, it might be useful to distinguish real-time datagrams (such as those used by an IP telephony application) from non-real-time traffic (for example, FTP). The specific level of service to be provided is a policy issue determined by the router's administrator. █ Datagram length. This is the total length of the IP datagram (header plus data), measured in bytes. Since this field is 16 bits long, the theoretical maximum size of the IP datagram is 65,535 bytes. However, datagrams are rarely larger than 1,500 bytes. █ Identifier, flags, fragmentation offset. These three fields have to do with so-called IP fragmentation, a topic we will consider in depth shortly. Interestingly, the new version of IP, IPv6, does not allow for fragmentation at routers. █ Time-to-live. The time-to-live (TTL) field is included to ensure that datagrams do not circulate forever (due to, for example, a long-lived routing loop) in the network. This field is decremented by one each time the datagram is processed by a router. If the TTL field reaches 0, the datagram must be dropped. █ Protocol. This field is used only when an IP datagram reaches its final destination. The value of this field indicates the specific transport-layer protocol to which the data portion of this IP datagram should be passed. For example, a value of 6 indicates that the data portion is passed to TCP, while a value of 17 indicates that the data is passed to UDP. █ Header checksum. The header checksum aids a router in detecting bit errors in a received IP datagram. The header checksum is computed by treating each 2 bytes in the header as a number and summing these numbers using 1s complement arithmetic. █ Source and destination IP addresses. When a source creates a datagram, it inserts its IP address into the source IP address field and inserts the address of the ultimate destination into the destination IP address field. █ Options. The options fields allow an IP header to be extended. Header options were meant to be used rarely—hence the decision to save overhead by not including the information in options fields in every datagram header. However, the mere existence of options does complicate matters—since datagram headers can be of variable length, one cannot determine a priori where the data field will start. Also, since some datagrams may require options processing and others may not, the amount of time needed to process an IP datagram at a router can vary greatly. These considerations become particularly important for IP processing in high-performance routers and hosts. For these reasons and others, IP options were dropped in the IPv6 header, as discussed in Section 4.4.4. █ Data (payload). Finally, we come to the last and most important field—the raison d'être for the datagram in the first place! In most circumstances, the data field of the IP datagram contains the transport-layer segment (TCP or UDP) to be delivered to the destination. However, the data field can carry other types of data, such as ICMP messages (discussed in Section 4.4.3).

Draw an IP datagram and identify what all the fields are for

each of the links along the route between sender and destination can use different link-layer protocols, and each of these protocols can have different MTUs (max transmission units - how much data it can hold). If an outgoing link has an MTU that is smaller than the length of the IP datagram then the datagram is broken 1 or more smaller datagrams known as fragments. http://i.stack.imgur.com/0XaEs.png

Explain IP datagram fragmentation

dual-stack approach: where IPv6 nodes also have a complete IPv4 implementation. Such a node, referred to as an IPv6/IPv4 node in RFC 4213, has the ability to send and receive both IPv4 and IPv6 datagrams. When interoperating with an IPv4 node, an IPv6/IPv4 node can use IPv4 datagrams; when interoperating with an IPv6 node, it can speak IPv6. IPv6/IPv4 nodes must have both IPv6 and IPv4 addresses. They must furthermore be able to determine whether another node is IPv6-capable or IPv4-only. This problem can be solved using the DNS (see Chapter 2), which can return an IPv6 address if the node name being resolved is IPv6-capable, or otherwise return an IPv4 address. Of course, if the node issuing the DNS request is only IPv4-capable, the DNS returns only an IPv4 address. In the dual-stack approach, if either the sender or the receiver is only IPv4- capable, an IPv4 datagram must be used. As a result, it is possible that two IPv6- capable nodes can end up, in essence, sending IPv4 datagrams to each other. must furthermore be able to determine whether another node is IPv6-capable or IPv4-only. This problem can be solved using the DNS (see Chapter 2), which can return an IPv6 address if the node name being resolved is IPv6-capable, or otherwise return an IPv4 address. Of course, if the node issuing the DNS request is only IPv4-capable, the DNS returns only an IPv4 address. █ tunneling - Suppose two IPv6 nodes want to interoperate using IPv6 datagrams but are connected to each other by intervening IPv4 routers. We refer to the intervening set of IPv4 routers between two IPv6 routers as a tunnel. With tunneling, the IPv6 node on the sending side of the tunnel takes the entire IPv6 datagram and puts it in the data (payload) field of an IPv4 datagram. The IPv6 node on the receiving side of the tunnel eventually receives the IPv4 datagram (it is the destination of the IPv4 datagram!), determines that the IPv4 datagram contains an IPv6 datagram, extracts the IPv6 datagram, and then routes the IPv6 datagram exactly as it would if it had received the IPv6 datagram from a directly connected IPv6 neighbor.

Explain the dual-stack vs. tunneling approaches for IPv4-to-IPv6 transition

The 802.11 standard requires that an AP periodically send beacon frames, each of which includes the AP's SSID and MAC address. Your wireless station, knowing that APs are sending out beacon frames, scans the 11 channels, seeking beacon frames from any APs that may be out there. To gain Internet access, your wireless station needs to join exactly one of the subnets and hence needs to associate with exactly one of the APs. Associating means the wireless station creates a virtual wire between itself and the AP. Specifically, only the associated AP will send data frames (that is, frames containing data, such as a datagram) to your wireless station, and your wireless station will send data frames into the Internet only through the associated AP. Once associated with an AP, the host will want to join the subnet to which the AP belongs. Thus, the host will typically send a DHCP discovery message into the subnet via the AP in order to obtain an IP address on the subnet. Once the address is obtained, the rest of the world then views that host simply as another host with an IP address in that subnet.

How are wifi connections established?

The role of the switch is to receive incoming link-layer frames and forward them onto outgoing links; we'll study this forwarding function in detail in this subsection. We'll see that the switch itself is transparent to the hosts and routers in the subnet; that is, a host/router addresses a frame to another host/router (rather than addressing the frame to the switch) and happily sends the frame into the LAN, unaware that a switch will be receiving the frame and forwarding it. The rate at which frames arrive to any one of the switch's output interfaces may temporarily exceed the link capacity of that interface. To accommodate this problem, switch output interfaces have buffers, in much the same way that router output interfaces have buffers for datagrams. Filtering is the switch function that determines whether a frame should be forwarded to some interface or should just be dropped. Forwarding is the switch function that determines the interfaces to which a frame should be directed, and then moves the frame to those interfaces. Switch filtering and forwarding are done with a switch table. The switch table contains entries for some, but not necessarily all, of the hosts and routers on a LAN. An entry in the switch table contains (1) a MAC address, (2) the switch interface that leads toward that MAC address, and (3) the time at which the entry was placed in the table. switches forward packets based on MAC addresses rather than on IP addresses. If there is no destination address in the switch table then the switch broadcasts the frame. A switch has the wonderful property (particularly for the already-overworked network administrator) that its table is built automatically, dynamically, and autonomously—without any intervention from a network administrator or from a configuration protocol. In other words, switches are self-learning. This capability is accomplished as follows: 1. The switch table is initially empty. 2. For each incoming frame received on an interface, the switch stores in its table (1) the MAC address in the frame's source address field, (2) the interface from which the frame arrived, and (3) the current time. In this manner the switch records in its table the LAN segment on which the sender resides. If every host in the LAN eventually sends a frame, then every host will eventually get recorded in the table. 3. The switch deletes an address in the table if no frames are received with that address as the source address after some period of time (the aging time). In this manner, if a PC is replaced by another PC (with a different adapter), the MAC address of the original PC will eventually be purged from the switch table.

How do switches work?

In the center-based approach to building a spanning tree, a center node (also known as a rendezvous point or a core) is defined. Nodes then unicast tree-join messages addressed to the center node. A tree-join message is forwarded using unicast routing toward the center until it either arrives at a node that already belongs to the spanning tree or arrives at the center. In either case, the path that the tree-join message has followed defines the branch of the spanning tree between the edge node that initiated the tree-join message and the center. One can think of this new path as being grafted onto the existing spanning tree. Figure 4.46 illustrates the construction of a center-based spanning tree. Suppose that node E is selected as the center of the tree. Suppose that node F first joins the tree and forwards a tree-join message to E. The single link EF becomes the initial spanning tree. Node B then joins the spanning tree by sending its tree-join message to E. Suppose that the unicast path route to E from B is via D. In this case, the tree-join message results in the path BDE being grafted onto the spanning tree. Node A next joins the spanning group by forwarding its tree-join message towards E. If A's unicast path to E is through B, then since B has already joined the spanning tree, the arrival of A's tree-join message at B will result in the AB link being immediately grafted onto the spanning tree. Node C joins the spanning tree next by forwarding its tree-join message directly to E. Finally, because the unicast routing from G to E must be via node D, when G sends its tree-join message to E, the GD link is grafted onto the spanning tree at node D. (image on pg 404)

How do you construct a broadcast spanning tree?

a router looks at the header value of a packet and uses this to determine which destination in the forwarding table to send the packet to. http://www.networkinginfoblog.com/contentsimages/Routing%20algorithms%20determine%20values%20in%20forwarding%20tables..JPG

How does a routers forwarding table work?

Output port processing takes packets that have been stored in the output port's memory and transmits them over the output link. This includes selecting and de-queueing packets for transmission, and performing the needed linklayer and physical-layer transmission functions.

How does a routers output processing work?

when data arrives on the input port the router uses the forwarding table to look up the output port to which an arriving packet will be forwarded via the switching fabric.

How does router input processing work?

queues can occur at both the input and output ports. When a router runs out of memory to store packets packet loss occurs. packet schedulers choose which packet in the queue should be sent (ie, first come first serve scheduling)

How does router queueing work?

The switching fabric is at the very heart of a router, as it is through this fabric that the packets are actually switched (that is, forwarded) from an input port to an output port. █ Switching via memory. The simplest, earliest routers were traditional computers, with switching between input and output ports being done under direct control of the CPU (routing processor). █ Switching via a bus. In this approach, an input port transfers a packet directly to the output port over a shared bus, without intervention by the routing processor. █ Switching via an interconnection network. unlike the previous two switching approaches, crossbar networks are capable of forwarding multiple packets in parallel. However, if two packets from two different input ports are destined to the same output port, then one will have to wait at the input, since only one packet can be sent over any given bus at a time.

How does router switching work and what are the 3 primary methods for switching?

if an incoming packet is waiting in a queue to enter an outgoing port the packet behind it must also wait. Figure 4.11 shows an example in which two packets (darkly shaded) at the front of their input queues are destined for the same upper-right output port. Suppose that the switch fabric chooses to transfer the packet from the front of the upper-left queue. In this case, the darkly shaded packet in the lower-left queue must wait. But not only must this darkly shaded packet wait, so too must the lightly shaded packet that is queued behind that packet in the lower-left queue, even though there is no contention for the middle-right output port (the destination for the lightly shaded packet). http://phoenix.goucher.edu/~kelliher/s2008/cs325/apr14img14.png

In a router what is head of line blocking?

some network architectures require routers along a chosen path from source to destination to handshake with eachother to setup state before packets can be transferred

In the network layer, what is a connection setup?

• Framing. Almost all link-layer protocols encapsulate each network-layer datagram within a link-layer frame before transmission over the link. A frame consists of a data field, in which the network-layer datagram is inserted, and a number of header fields. The structure of the frame is specified by the link-layer protocol. • Link access. Amedium access control (MAC) protocol specifies the rules by which a frame is transmitted onto the link. For point-to-point links that have a single sender at one end of the link and a single receiver at the other end of the link, the MAC protocol is simple (or nonexistent)—the sender can send a frame whenever the link is idle. The more interesting case is when multiple nodes share a single broadcast link—the so-called multiple access problem. Here, the MAC protocol serves to coordinate the frame transmissions of the many nodes. • Reliable delivery. When a link-layer protocol provides reliable delivery service, it guarantees to move each network-layer datagram across the link without error. A link-layer reliable delivery service is often used for links that are prone to high error rates, such as a wireless link, with the goal of correcting an error locally—on the link where the error occurs—rather than forcing an end-to end retransmission of the data by a transport- or application-layer protocol. However, link-layer reliable delivery can be considered an unnecessary overhead for low bit-error links, including fiber, coax, and many twisted-pair copper links. For this reason, many wired link-layer protocols do not provide a reliable delivery service. • Error detection and correction. The link-layer hardware in a receiving node can incorrectly decide that a bit in a frame is zero when it was transmitted as a one, and vice versa. Such bit errors are introduced by signal attenuation and electromagnetic noise. Because there is no need to forward a datagram that has an error, many link-layer protocols provide a mechanism to detect such bit errors. This is done by having the transmitting node include error-detection bits in the frame, and having the receiving node perform an error check.

What are the link layer services?

First consider the pros and cons of switches. As mentioned above, switches are plug-and-play, a property that is cherished by all the overworked network administrators of the world. Switches can also have relatively high filtering and forwarding rates—as shown in Figure 5.24, switches have to process frames only up through layer 2, whereas routers have to process datagrams up through layer 3. On the other hand, to prevent the cycling of broadcast frames, the active topology of a switched network is restricted to a spanning tree. Also, a large switched network would require large ARP tables in the hosts and routers and would generate substantial ARP traffic and processing. Furthermore, switches are susceptible to broadcast storms—if one host goes haywire and transmits an endless stream of Ethernet broadcast frames, the switches will forward all of these frames, causing the entire network to collapse. Now consider the pros and cons of routers. Because network addressing is often hierarchical (and not flat, as is MAC addressing), packets do not normally cycle through routers even when the network has redundant paths. (However, packets can cycle when router tables are misconfigured; but as we learned in Chapter 4, IP uses a special datagram header field to limit the cycling.) Thus, packets are not restricted to a spanning tree and can use the best path between source and destination. Because routers do not have the spanning tree restriction, they have allowed the Internet to be built with a rich topology that includes, for example, multiple active links between Europe and North America. Another feature of routers is that they provide firewall protection against layer-2 broadcast storms. Perhaps the most significant drawback of routers, though, is that they are not plug-and-play—they and the hosts that connect to them need their IP addresses to be configured. Also, routers often have a larger per-packet processing time than switches, because they have to process up through the layer-3 fields.

What are the pros and cons of switches vs routers?

guaranteed minimum bandwidth, no loss guarantee, in order, no timing, congestion indication

What does the ATM ABR service model offer?

guaranteed constant rate bandwidth, loss guarantee, in order, timing maintained, no congestion

What does the ATM CBR service model offer?

best effort: no bandwidth guarantee, no loss guarantee, any order possible, no timing guarantee, no congestion indication

What does the internet network service model offer?

Because there are both network-layer addresses (for example, Internet IP addresses) and link-layer addresses (that is, MAC addresses), there is a need to translate between them. For the Internet, this is the job of the Address Resolution Protocol (ARP). A sending adapter will construct a linklayer frame containing the destination's MAC address and send the frame into the LAN. An ARP module in the sending host takes any IP address on the same LAN as input, and returns the corresponding MAC address. So we see that ARP resolves an IP address to a MAC address. In many ways it is analogous to DNS (studied in Section 2.5), which resolves host names to IP addresses. However, one important difference between the two resolvers is that DNS resolves host names for hosts anywhere in the Internet, whereas ARP resolves IP addresses only for hosts and router interfaces on the same subnet. Each host and router has an ARP table in its memory, which contains mappings of IP addresses to MAC addresses. If a host wants to send a datagram that is not in the ARP table it broadcasts an ARP query packet to all other hosts and routers on the subnet. The frame containing the ARP query is received by all the other adapters on the subnet, and (because of the broadcast address) each adapter passes the ARP packet within the frame up to its ARP module. Each of these ARP modules checks to see if its IP address matches the destination IP address in the ARP packet. The one with a match sends back to the querying host a response ARP packet with the desired mapping. The querying host can then update its ARP table and send its IP datagram, encapsulated in a link-layer frame whose destination MAC is that of the host or router responding to the earlier ARP query.

What is ARP?

In BGP, an autonomous system is identified by its globally unique autonomous system number (ASN). AS numbers, like IP addresses, are assigned by ICANN regional registries. When a router advertises a prefix across a BGP session, it includes with the prefix a number of BGP attributes. of the more important attributes are AS-PATH and NEXT-HOP. Providing the critical link between the inter-AS and intra-AS routing protocols, the NEXT-HOP attribute has a subtle but important use. The NEXT-HOP is the router interface that begins the AS-PATH. To gain insight into this attribute, let's again refer to Figure 4.40. Consider what happens when the gateway router 3a in AS3 advertises a route to gateway router 1c in AS1 using eBGP. The route includes the advertised prefix, which we'll call x, and an AS-PATH to the prefix. This advertisement also includes the NEXT-HOP, which is the IP address of the router 3a interface that leads to 1c. (Recall that a router has multiple IP addresses, one for each of its interfaces.) Now consider what happens when router 1d learns about this route from iBGP. After learning about this route to x, router 1d may want to forward packets to x along the route, that is, router 1d may want to include the entry (x, l) in its forwarding table, where l is its interface that begins the least-cost path from 1d towards the gateway router 1c. To determine l, 1d provides the IP address in the NEXT-HOP attribute to its intra-AS routing module. Note that the intra-AS routing algorithm has determined the least-cost path to all subnets attached to the routers in AS1, including to the subnet for the link between 1c and 3a. From this least-cost path from 1d to the 1c-3a subnet, 1d determines its router interface l that begins this path and then adds the entry (x, l) to its forwarding table. Whew! In summary, the NEXT-HOP attribute is used by routers to properly configure their forwarding tables. https://blog.thousandeyes.com/wp-content/uploads/2014/03/image01.png Figure 4.41 (pg 395) illustrates another situation where the NEXT-HOP is needed. In this figure, AS1 and AS2 are connected by two peering links. A router in AS1 could learn about two different routes to the same prefix x. These two routes could have the same AS-PATH to x, but could have different NEXT-HOP values corresponding to the different peering links. Using the NEXT-HOP values and the intra-AS routing algorithm, the router can determine the cost of the path to each peering link, and then apply hot-potato routing (see Section 4.5.3) to determine the appropriate interface.

What is ASN, AS-PATH, and NEXT-HOP?

BGP is the standard inter-AS routing protocol used in today's internet. BGP provides each AS a means to 1. Obtain subnet reachability information from neighboring ASs. 2. Propagate the reachability information to all routers internal to the AS. 3. Determine "good" routes to subnets based on the reachability information and on AS policy. Most importantly, BGP allows each subnet to advertise its existence to the rest of the Internet. A subnet screams "I exist and I am here," and BGP makes sure that all the ASs in the Internet know about the subnet and how to get there. If it weren't for BGP, each subnet would be isolated—alone and unknown by the rest of the Internet. In BGP, pairs of routers exchange routing information over semipermanent TCP connections. There is typically one such BGP TCP connection for each link that directly connects two routers in two different ASs. For each TCP connection, the two routers at the end of the connection are called BGP peers, and the TCP connection along with all the BGP messages sent over the connection is called a BGP session. Furthermore, a BGP session that spans two ASs is called an external BGP (eBGP) session, and a BGP session between routers in the same AS is called an internal BGP (iBGP) session. https://blog.thousandeyes.com/wp-content/uploads/2014/03/image01.png

What is BGP?

Classless Interdomain Routing (CIDR—pronounced cider). The x most significant bits of an address of the form a.b.c.d/x constitute the network portion of the IP address, and are often referred to as the prefix (or network prefix) of the address. An organization is typically assigned a block of contiguous addresses, that is, a range of addresses with a common prefix. only these x leading prefix bits are considered by routers outside the organization's network. That is, when a router outside the organization forwards a datagram whose destination address is inside the organization, only the leading x bits of the address need be considered. This considerably reduces the size of the forwarding table in these routers, since a single entry of the form a.b.c.d/x will be sufficient to forward packets to any destination within the organization. The remaining 32-x bits of an address can be thought of as distinguishing among the devices within the organization, all of which have the same network prefix.

What is CIDR?

Dynamic Host Configuration Protocol (DHCP). Allows a host to obtain an IP address automatically without requiring it to be manually assigned by an administrator. A network administrator can configure DHCP so that a given host receives the same IP address each time it connects to the network, or a host may be assigned a temporary IP address that will be different each time the host connects to the network. In addition to host IP address assignment, DHCP also allows a host to learn additional information, such as its subnet mask, the address of its first-hop router (often called the default gateway), and the address of its local DNS server. http://www.dcs.bbk.ac.uk/~ptw/teaching/IWT/network-layer/dhcp-interaction.gif DHCP server discovery. The first task of a newly arriving host is to find a DHCP server with which to interact. This is done using a DHCP discover message. the DHCP client creates an IP datagram containing its DHCP discover message along with the broadcast destination IP address of 255.255.255.255 and a "this host" source IP address of 0.0.0.0. The DHCP client passes the IP datagram to the link layer, which then broadcasts this frame to all nodes attached to the subnet. DHCP server offer(s). A DHCP server receiving a DHCP discover message responds to the client with a DHCP offer message that is broadcast to all nodes on the subnet, again using the IP broadcast address of 255.255.255.255. Each server offer message contains the transaction ID of the received discover message, the proposed IP address for the client, the network mask, and an IP address lease time—the amount of time for which the IP address will be valid. DHCP request. The newly arriving client will choose from among one or more server offers and respond to its selected offer with a DHCP request message, echoing back the configuration parameters. • DHCP ACK. The server responds to the DHCP request message with a DHCP ACK message, confirming the requested parameters.

What is DHCP?

Internet Control Message Protocol (ICMP) is used by hosts and routers to communicate network- layer information to each other. The most typical use of ICMP is for error reporting. For example, when running a Telnet, FTP, or HTTP session, you may have encountered an error message such as "Destination network unreachable." ICMP messages are carried inside IP datagrams. Ping and traceroute are both implemented using ICMP.

What is ICMP

an intra-AS routing protocol that is very similar to OSPF

What is IS-IS?

MPLS is a scalable, protocol-independent transport. In an MPLS network, data packets are assigned labels. Packet-forwarding decisions are made solely on the contents of this label, without the need to examine the packet itself. Multiprotocol Label Switching (MPLS) network. Unlike the circuit-switched telephone network, MPLS is a packet-switched, virtual-circuit network in its own right. It has its own packet formats and forwarding behaviors. An MPLS-capable router is often referred to as a label-switched router, since it forwards an MPLS frame by looking up the MPLS label in its forwarding table and then immediately passing the datagram to the appropriate output interface. Thus, the MPLS-capable router need not extract the destination IP address and perform a lookup of the longest prefix match in the forwarding table. The true advantages of MPLS and the reason for current interest in MPLS, however, lie not in the potential increases in switching speeds, but rather in the new traffic management capabilities that MPLS enables. Note that there are two MLPS paths. IP routing protocols we studied in Chapter 4 would specify only a single, least-cost path to A. Thus, MPLS provides the ability to forward packets along routes that would not be possible using standard IP routing protocols. This allows a network operator can override normal IP routing and force some of the traffic headed toward a given destination along one path, and other traffic destined toward the same destination along another path (whether for policy, performance, or some other reason). It is also possible to use MPLS for many other purposes as well. It can be used to perform fast restoration of MPLS forwarding paths, e.g., to reroute traffic over a precomputed failover path in response to link failure [Kar 2000; Huang 2002; RFC 3469]. Finally, we note that MPLS can, and has, been used to implement so-called virtual private networks (VPNs). In implementing a VPN for a customer, an ISP uses its MPLS-enabled network to connect together the customer's various networks. MPLS can be used to isolate both the resources and addressing used by the customer's VPN from that of other users crossing the ISP's network

What is MPLS?

NAT allows many connected devices on a home network to interact with the outside network. The NAT-enabled router does not look like a router to the outside world. Instead the NAT router behaves to the outside world as a single device with a single IP address. In essence, the NAT-enabled router is hiding the details of the home network from the outside world. The router knows which host on the home network to send data to by using a NAT translation table. http://www.dcs.bbk.ac.uk/~ptw/teaching/IWT/network-layer/nat.gif

What is NAT?

OSPF was conceived as the successor to RIP and as such has a number of advanced features. At its heart, however, OSPF is a link-state protocol that uses flooding of link-state information and a Dijkstra least-cost path algorithm. With OSPF, a router constructs a complete topological map (that is, a graph) of the entire autonomous system. The router then locally runs Dijkstra's shortest-path algorithm to determine a shortest-path tree to all subnets, with itself as the root node. With OSPF, a router broadcasts routing information to all other routers in the autonomous system, not just to its neighboring routers. A router broadcasts linkstate information whenever there is a change in a link's state (for example, a change in cost or a change in up/down status). It also broadcasts a link's state periodically (at least once every 30 minutes), even if the link's state has not changed. An OSPF autonomous system can be configured hierarchically into areas. Each area runs its own OSPF link-state routing algorithm, with each router in an area broadcasting its link state to all other routers in that area. Within each area, one or more area border routers are responsible for routing packets outside the area. Lastly, exactly one OSPF area in the AS is configured to be the backbone area. The primary role of the backbone area is to route traffic between the other areas in the AS. The backbone always contains all area border routers in the AS and may contain nonborder routers as well. Inter-area routing within the AS requires that the packet be first routed to an area border router (intra-area routing), then routed through the backbone to the area border router that is in the destination area, and then routed to the final destination.

What is OSPF?

In computer networking, Point-to-Point Protocol (PPP) is a data link protocol used to establish a direct connection between two nodes. It can provide connection authentication, transmission encryption (using ECP, RFC 1968), and compression. PPP is used over many types of physical networks including serial cable, phone line, trunk line, cellular telephone, specialized radio links, and fiber optic links such as SONET. PPP is also used over Internet access connections. Internet service providers (ISPs) have used PPP for customer dial-up access to the Internet, since IP packets cannot be transmitted over a modem line on their own, without some data link protocol. Two derivatives of PPP, Point-to-Point Protocol over Ethernet (PPPoE) and Point-to-Point Protocol over ATM (PPPoA), are used most commonly by Internet Service Providers (ISPs) to establish a Digital Subscriber Line (DSL) Internet service connection with customers. PPP is commonly used as a data link layer protocol for connection over synchronous and asynchronous circuits, where it has largely superseded the older Serial Line Internet Protocol (SLIP) and telephone company mandated standards (such as Link Access Protocol, Balanced (LAPB) in the X.25 protocol suite). The only requirement for PPP is that the circuit provided be duplex. PPP was designed to work with numerous network layer protocols, including Internet Protocol (IP), TRILL, Novell's Internetwork Packet Exchange (IPX), NBF, DECnet and AppleTalk.

What is PPP?

RIP is a distance-vector protocol that operates in a manner very close to the idealized DV protocol we examined in Section 4.5.2. The version of RIP specified in RFC 1058 uses hop count as a cost metric; that is, each link has a cost of 1. The routing protocol is used in intra-AS setups. The maximum cost of a path is limited to 15, thus limiting the use of RIP to autonomous systems that are fewer than 15 hops in diameter. Recall that in DV protocols, neighboring routers exchange distance vectors with each other. The distance vector for any one router is the current estimate of the shortest path distances from that router to the subnets in the AS. In RIP, routing updates are exchanged between neighbors approximately every 30 seconds using a RIP response message. The response message sent by a router or host contains a list of up to 25 destination subnets within the AS, as well as the sender's distance to each of those subnets. Response messages are also known as RIP advertisements.

What is RIP?

The signal-to-noise ratio (SNR) is a relative measure of the strength of the received signal (i.e., the information being transmitted) and this noise. The SNR is typically measured in units of decibels (dB). a larger SNR makes it easier for the receiver to extract the transmitted signal from the background noise. ------------------------------------------ bit error rate (BER)— is roughly speaking, the probability that a transmitted bit is received in error at the receiver. the higher the SNR, the lower the BER.

What is SNR and BER?

In truth, it is not hosts and routers that have MAC addresses but rather their adapters (that is, network interfaces) that have MAC addresses. When an adapter wants to send a frame to some destination adapter, the sending adapter inserts the destination adapter's MAC address into the frame and then sends the frame into the LAN. a switch occassionally broadcasts an incoming frame onto all of its interfaces. We'll see in Chapter 6 that 802.11 also broadcasts frames. Thus, an adapter may receive a frame that isn't addressed to it. Thus, when an adapter receives a frame, it will check to see whether the destination MAC address in the frame matches its own MAC address. If there is a match, the adapter extracts the enclosed datagram and passes the datagram up the protocol stack. If there isn't a match, the adapter discards the frame, without passing the network-layer datagram up. Thus, the destination only will be interrupted when the frame is received. However, sometimes a sending adapter does want all the other adapters on the LAN to receive and process the frame it is about to send. In this case, the sending adapter inserts a special MAC broadcast address into the destination address field of the frame. For LANs that use 6-byte addresses (such as Ethernet and 802.11), the broadcast address is a string of 48 consecutive 1s (that is, FF-FF-FF-FF-FFFF in hexadecimal notation).

What is a MAC address?

a switch that supports VLANs allows multiple virtual local area networks to be defined over a single physical local area network infrastructure. Hosts within a VLAN communicate with each other as if they (and no other hosts) were connected to the switch. In a port-based VLAN, the switch's ports (interfaces) are divided into groups by the network manager. Each group constitutes a VLAN, with the ports in each VLAN forming a broadcast domain (i.e., broadcast traffic from one port can only reach other ports in the group). VLANs isolate traffic, limiting broadcast traffic to many users improves LAN performance. VLANs use switches more efficiently as you can virtually create a group rather than separate each group with a switch. VLANs are also good for managing users, ie put regular employees and managers in different groups.

What is a VLAN?

in the Internet architecture (and other network architectures such as ATM [Black 1995]), a multicast packet is addressed using address indirection. That is, a single identifier is used for the group of receivers, and a copy of the packet that is addressed to the group using this single identifier is delivered to all of the multicast receivers associated with that group. In the Internet, the single identifier that represents a group of receivers is a class D multicast IP address. The group of receivers associated with a class D address is referred to as a multicast group. The first multicast routing protocol used in the Internet was the Distance-Vector Multicast Routing Protocol (DVMRP) [RFC 1075]. DVMRP implements source-based trees with reverse path forwarding and pruning. DVMRP uses an RPF algorithm with pruning, as discussed above. Perhaps the most widely used Internet multicast routing protocol is the Protocol-Independent Multicast (PIM) routing protocol, which explicitly recognizes two multicast distribution scenarios. In dense mode [RFC 3973], multicast group members are densely located; that is, many or most of the routers in the area need to be involved in routing multicast datagrams. PIM dense mode is a flood-and-prune reverse path forwarding technique similar in spirit to DVMRP. In sparse mode [RFC 4601], the number of routers with attached group members is small with respect to the total number of routers; group members are widely dispersed. PIM sparse mode uses rendezvous points to set up the multicast distribution tree. In source-specific multicast (SSM) [RFC 3569, RFC 4607], only a single sender is allowed to send traffic into the multicast tree, considerably simplifying tree construction and maintenance.

What is a multicast group and what are the two multicast routing algorithms?

a router's control functions—executing the routing protocols, responding to attached links that go up or down, and performing management functions. These router control plane functions are usually implemented in software and execute on the routing processor (typically a traditional CPU).

What is a router control plane?

A router's input ports, output ports, and switching fabric together implement the forwarding function. The switching fabric connects the router's input ports to its output ports. http://www.cs.umd.edu/~shankar/417-F01/Slides/chapter4b-aus/img023.gif

What is a router forwarding plane?

In IP terms, a network interconnecting three host interfaces and one router interface forms a subnet [RFC 950]. (A subnet is also called an IP network or simply a network in the Internet literature.) IP addressing assigns an address to this subnet: 223.1.1.0/24, where the /24 notation, sometimes known as a subnet mask, indicates that the leftmost 24 bits of the 32-bit quantity define the subnet address. The subnet 223.1.1.0/24 thus consists of the three host interfaces (223.1.1.1, 223.1.1.2, and 223.1.1.3) and one router interface (223.1.1.4). Any additional hosts attached to the 223.1.1.0/24 subnet would be required to have an address of the form 223.1.1.xxx.

What is a subnet?

A host typically has only a single link into the network; when IP in the host wants to send a datagram, it does so over this link. The boundary between the host and the physical link is called an interface. Now consider a router and its interfaces. Because a router's job is to receive a datagram on one link and forward the datagram on some other link, a router necessarily has two or more links to which it is connected. The boundary between the router and any one of its links is also called an interface. A router thus has multiple interfaces, one for each of its links. Because every host and router is capable of sending and receiving IP datagrams, IP requires each host and router interface to have its own IP address. Thus, an IP address is technically associated with an interface, rather than with the host or router containing that interface.

What is an interface?

a block of ip addresses constrained to be 8, 16, or 24 bits in length. This became problematic because /24 class C = 254 address which is not enough for most organizations but /16 = 65,634 is too much. So organizations that needed say 2,000 were allocated /16. This led to poor utilization of assigned ip addresses.

What is classful addressing?

A new protocol was needed when it was realized that ipv4 would run out of ip addresses. Improvements: █ more address - ipv6 features 128 bit addresses. IPv6 has introduced a new type of address, called an anycast address, which allows a datagram to be delivered to any one of a group of hosts. (This feature could be used, for example, to send an HTTP GET to the nearest of a number of mirror sites that contain a given document.) █ A streamlined 40-byte header. As discussed below, a number of IPv4 fields have been dropped or made optional. The resulting 40-byte fixed-length header allows for faster processing of the IP datagram. A new encoding of options allows for more flexible options processing. █ Flow labeling and priority. "labeling of packets belonging to particular flows for which the sender requests special handling, such as a nondefault quality of service or real-time service." For example, audio and video transmission might likely be treated as a flow. On the other hand, the more traditional applications, such as file transfer and e-mail, might not be treated as flows. It is possible that the traffic carried by a high-priority user (for example, someone paying for better service for their traffic) might also be treated as a flow. What is clear, however, is that the designers of IPv6 foresee the eventual need to be able to differentiate among the flows, even if the exact meaning of a flow has not yet been determined.

What is ipv6 and what are its improvements?

A style of forwarding table in a datagram network. With this style of forwarding table, the router matches a prefix of the packet's destination address with the entries in the table; if there's a match, the router forwards the packet to a link associated with the match. Use the LONGEST prefix that matches the destination address. http://image.slidesharecdn.com/chapter4v6-140114160530-phpapp01/95/chapter-4-v611-19-638.jpg?cb=1389737323

What is prefix and prefix matching?

The Bellman-Ford algorithm does not prevent routing loops from happening and suffers from the count-to-infinity problem. The core of the count-to-infinity problem is that if A tells B that it has a path somewhere, there is no way for B to know if the path has B as a part of it. To see the problem clearly, imagine a subnet connected like A-B-C-D-E-F, and let the metric between the routers be "number of jumps". Now suppose that A is taken offline. In the vector-update-process B notices that the route to A, which was distance 1, is down - B does not receive the vector update from A. The problem is, B also gets an update from C, and C is still not aware of the fact that A is down - so it tells B that A is only two jumps from C (C to B to A), which is false. Since B doesn't know that the path from C to A is through itself (B), it updates its table with the new value "B to A = 2 + 1". Later on, B forwards the update to C and due to the fact that A is reachable through B (From C point of view), C decides to update its table to "C to A = 3 + 1". This slowly propagates through the network until it reaches infinity (in which case the algorithm corrects itself, due to the relaxation property of Bellman-Ford). See http://cseweb.ucsd.edu/classes/fa11/cse123-a/123f11_Lec9.pdf for poison reverse example

What is the count-to-infinity problem and how does poison reverse help solve it?

By the late 1990s, most companies and universities had replaced their LANs with Ethernet installations using a hub-based star topology. In such an installation the hosts (and routers) are directly connected to a hub with twisted-pair copper wire. A hub is a physical-layer device that acts on individual bits rather than frames. When a bit, representing a zero or a one, arrives from one interface, the hub simply re-creates the bit, boosts its energy strength, and transmits the bit onto all the other interfaces. Thus, Ethernet with a hub-based star topology is also a broadcast LAN—whenever a hub receives a bit from one of its interfaces, it sends a copy out on all of its other interfaces. In particular, if a hub receives frames from two different interfaces at the same time, a collision occurs and the nodes that created the frames must retransmit. In the early 2000s Ethernet experienced yet another major evolutionary change. Ethernet installations continued to use a star topology, but the hub at the center was replaced with a switch. We'll be examining switched Ethernet in depth later in this chapter. For now, we only mention that a switch is not only "collision-less" but is also a bona-fide store-and-forward packet switch; but unlike routers, which operate up through layer 3, a switch operates only up through layer 2.

What is the difference between a router, switch, and a hub?

forwarding is the act of forwarding packet data from one location to another. routing is determining the path that the packet takes

What is the difference between forwarding and routing?

Computer networks that provide only a connection service at the network layer are called virtual-circuit (VC) networks; computer networks that provide only a connectionless service at the network layer are called datagram networks. The internet is a datagram network. ie - similar to tcp and udp, connectionless requires no handshake

What is the difference between virtual circuit and datagram networks?

As the number of routers becomes large, the overhead involved in computing, storing, and communicating routing information involved with routing algorithms becomes prohibitive. Additionally, an organization should be able to run and administer its network as it wishes, while still being able to connect its network to other outside networks. Both of these problems can be solved by organizing routers into autonomous systems (ASs), with each AS consisting of a group of routers that are typically under the same administrative control (e.g., operated by the same ISP or belonging to the same company network). Routers within the same AS all run the same routing algorithm (for example, an LS or DV algorithm) and have information about each other—exactly as was the case in our idealized model in the preceding section. The routing algorithm running within an autonomous system is called an intraautonomous system routing protocol. It will be necessary, of course, to connect ASs to each other, and thus one or more of the routers in an AS will have the added task of being responsible for forwarding packets to destinations outside the AS; these routers are called gateway routers. Figure 4.32 provides a simple example with three ASs: AS1, AS2, and AS3. In this figure, the heavy lines represent direct link connections between pairs of routers. The thinner lines hanging from the routers represent subnets that are directly connected to the routers. AS1 has four routers—1a, 1b, 1c, and 1d— which run the intra-AS routing protocol used within AS1. Thus, each of these four routers knows how to forward packets along the optimal path to any destination within AS1. Similarly, autonomous systems AS2 and AS3 each have three routers. Note that the intra-AS routing protocols running in AS1, AS2, and AS3 need not be the same. Also note that the routers 1b, 1c, 2a, and 3a are all gateway routers. If an AS has more than one gateway then the problem of forwarding becomes more challenging. For example, consider a router in AS1 and suppose it receives a packet whose destination is outside the AS. The router should clearly forward the packet to one of its two gateway routers, 1b or 1c, but which one? To solve this problem, AS1 needs (1) to learn which destinations are reachable via AS2 and which destinations are reachable via AS3, and (2) to propagate this reachability information to all the routers within AS1, so that each router can configure its forwarding table to handle external-AS destinations. These two tasks—obtaining reachability information from neighboring ASs and propagating the reachability information to all routers internal to the AS—are handled by the inter-AS routing protocol. Since the inter-AS routing protocol involves communication between two ASs, the two communicating ASs must run the same inter-AS routing protocol. In fact, in the Internet all ASs run the same inter-AS routing protocol, called BGP4 http://i.imgur.com/H6FsInY.png

explain hierarchical routing

http://www.dibyahash.in/CN2/pic5.jpg

how do parity checks work?

see page 443

how does the CRC checksum scheme work?

polling protocol. The polling protocol requires one of the nodes to be designated as a master node. The master node polls each of the nodes in a round-robin fashion. In particular, the master node first sends a message to node 1, saying that it (node 1) can transmit up to some maximum number of frames. After node 1 transmits some frames, the master node tells node 2 it (node 2) can transmit up to the maximum number of frames. (The master node can determine when a node has finished sending its frames by observing the lack of a signal on the channel.) The procedure continues in this manner, with the master node polling each of the nodes in a cyclic manner. The polling protocol eliminates the collisions and empty slots that plague random access protocols. ---------------------------------- token-passing protocol. In this protocol there is no master node. Asmall, special-purpose frame known as a token is exchanged among the nodes in some fixed order. For example, node 1 might always send the token to node 2, node 2 might always send the token to node 3, and node N might always send the token to node 1. When a node receives a token, it holds onto the token only if it has some frames to transmit; otherwise, it immediately forwards the token to the next node. If a node does have frames to transmit when it receives the token, it sends up to a maximum number of frames and then forwards the token to the next node. Token passing is decentralized and highly efficient. But it has its problems as well. For example, the failure of one node can crash the entire channel. Or if a node accidentally neglects to release the token, then some recovery procedure must be invoked to get the token back in circulation.

what are the 2 taking turns protocols and how do they work?

channel partitioning protocols, random access protocols, and taking-turns protocols.

what are the 3 categories of multiple access protocols?

TDM. As an example, suppose the channel supports N nodes and that the transmission rate of the channel is R bps. TDM divides time into time frames and further divides each time frame into N time slots. Returning to our cocktail party analogy, a TDM-regulated cocktail party would allow one partygoer to speak for a fixed period of time, then allow another partygoer to speak for the same amount of time, and so on. Once everyone had had a chance to talk, the pattern would repeat. it has two major drawbacks. First, a node is limited to an average rate of R/N bps even when it is the only node with packets to send. A second drawback is that a node must always wait for its turn in the transmission sequence—again, even when it is the only node with a frame to send. Imagine the partygoer who is the only one with anything to say (and imagine that this is the even rarer circumstance where everyone wants to hear what that one person has to say). Clearly, TDM would be a poor choice for a multiple access protocol for this particular party. ------------------------ FDM. While TDM shares the broadcast channel in time, FDM divides the R bps channel into different frequencies (each with a bandwidth of R/N) and assigns each frequency to one of the N nodes. FDM thus creates N smaller channels of R/N bps out of the single, larger R bps channel. FDM shares both the advantages and drawbacks of TDM. It avoids collisions and divides the bandwidth fairly among the N nodes. However, FDM also shares a principal disadvantage with TDM—a node is limited to a bandwidth of R/N, even when it is the only node with packets to send. ------------------------ http://www.dee.ufma.br/~cbrandao/disciplinas/teoria/EE5085-1/ebook/LivroKurose_primeiraedicao/5.3_arquivos/deluxe-content_arquivos/05-11.jpe ------------------------ A third channel partitioning protocol is code division multiple access (CDMA). While TDM and FDM assign time slots and frequencies, respectively, to the nodes, CDMA assigns a different code to each node. Each node then uses its unique code to encode the data bits it sends. If the codes are chosen carefully, CDMA networks have the wonderful property that different nodes can transmit simultaneously and yet have their respective receivers correctly receive a sender's encoded data bits (assuming the receiver knows the sender's code) in spite of interfering transmissions by other nodes.

what are the 3 channel partitioning protocols and how do they work?

• Decreasing signal strength. Electromagnetic radiation attenuates as it passes through matter (e.g., a radio signal passing through a wall). Even in free space, the signal will disperse, resulting in decreased signal strength (sometimes referred to as path loss) as the distance between sender and receiver increases.------------------------------------------ • Interference from other sources. Radio sources transmitting in the same frequency band will interfere with each other. For example, 2.4 GHz wireless phones and 802.11b wireless LANs transmit in the same frequency band. Thus, the 802.11b wireless LAN user talking on a 2.4 GHz wireless phone can expect that neither the network nor the phone will perform particularly well. In addition to interference from transmitting sources, electromagnetic noise within the environment (e.g., a nearby motor, a microwave) can result in interference.------------------------------------------ • Multipath propagation. Multipath propagation occurs when portions of the electromagnetic wave reflect off objects and the ground, taking paths of different lengths between a sender and receiver. This results in the blurring of the received signal at the receiver. Moving objects between the sender and receiver can cause multipath propagation to change over time.

what are the 3 important differences between a wired link and a wireless link?

In a random access protocol, a transmitting node always transmits at the full rate of the channel, namely, R bps. When there is a collision, each node involved in the collision repeatedly retransmits its frame (that is, packet) until its frame gets through without a collision. But when a node experiences a collision, it doesn't necessarily retransmit the frame right away. Instead it waits a random delay before retransmitting the frame. ----------------------------- slotted ALOHA - The nodes are synchronized so that each node knows when the slots begin. If there is a collision, the node detects the collision before the end of the slot. The node retransmits its frame in each subsequent slot with probability p until the frame is transmitted without a collision. Problem is efficiency, collisions can result in empty wasted slots. http://cfs16.tistory.com/image/1/tistory/2010/12/01/13/00/4cf5c84e0e9e6 ----------------------------- "regular" ALOHA - In pure ALOHA, when a frame first arrives (that is, a network-layer datagram is passed down from the network layer at the sending node), the node immediately transmits the frame in its entirety into the broadcast channel. If a transmitted frame experiences a collision with one or more other transmissions, the node will then immediately (after completely transmitting its collided frame) retransmit the frame with probability p. Otherwise, the node waits for a frame transmission time. After this wait, it then transmits the frame with probability p, or waits (remaining idle) for another frame time with probability 1 - p. ----------------------------- CSMA/CD - follows these conversation rules: • Listen before speaking. If someone else is speaking, wait until they are finished. In the networking world, this is called carrier sensing—a node listens to the channel before transmitting. If a frame from another node is currently being transmitted into the channel, a node then waits until it detects no transmissions for a short amount of time and then begins transmission. • If someone else begins talking at the same time, stop talking. In the networking world, this is called collision detection—a transmitting node listens to the channel while it is transmitting. If it detects that another node is transmitting an interfering ----------------------------- binary exponential backoff - used in ethernet. when transmitting a frame that has already experienced n collisions, a node chooses the value of K at random from {0, 1, 2, . . . . 2^(n-1)}. Thus, the more collisions experienced by a frame, the larger the interval from which K is chosen.

what are the 4 random access protocols and how do they work?

802.11b 2.4-2.485 GHz up to 11 Mbps - 2.4-2.485 GHz is an unlicensed frequency band that competes with 2.4 GHz phones and microwave ovens ------------------------------- 802.11a 5.1-5.8 GHz up to 54 Mbps - By operating at a higher frequency, 802.11a LANs have a shorter transmission distance for a given power level and suffer more from multipath propagation. ------------------------------- 802.11g 2.4-2.485 GHz up to 54 Mbps -------------------------------

what are the different wifi channels and frequencies?

ATM is a core protocol used over the SONET/SDH backbone of the public switched telephone network (PSTN) and Integrated Services Digital Network (ISDN), but its use is declining in favour of all IP. ATM provides functionality that is similar to both circuit switching and packet switching networks: ATM uses asynchronous time-division multiplexing,[4][5] and encodes data into small, fixed-sized packets (ISO-OSI frames) called cells. This differs from approaches such as the Internet Protocol or Ethernet that use variable sized packets and frames. ATM uses a connection-oriented model in which a virtual circuit must be established between two endpoints before the actual data exchange begins.[5] These virtual circuits may be "permanent", i.e. dedicated connections that are usually preconfigured by the service provider, or "switched", i.e. set up on a per-call basis using signalling and disconnected when the call is terminated. ATM eventually became dominated by Internet Protocol (IP) only technology (and Wireless or Mobile ATM never got any foothold).

what is ATM?

carrier sense multiple access (meaning that each station senses the channel before transmitting, and refrains from transmitting when the channel is sensed busy) with collision avoidance ------------------------------------------ Suppose that a station (wireless station or an AP) has a frame to transmit. 1. If initially the station senses the channel idle, it transmits its frame after a short period of time known as the Distributed Inter-frame Space (DIFS); see Figure 6.10. 2. Otherwise, the station chooses a random backoff value using binary exponential backoff (as we encountered in Section 5.3.2) and counts down this value when the channel is sensed idle. While the channel is sensed busy, the counter value remains frozen. 3. When the counter reaches zero (note that this can only occur while the channel is sensed idle), the station transmits the entire frame and then waits for an acknowledgment. 4. If an acknowledgment is received, the transmitting station knows that its frame has been correctly received at the destination station. If the station has another frame to send, it begins the CSMA/CA protocol at step 2. If the acknowledgment isn't received, the transmitting station reenters the backoff phase in step 2, with the random value chosen from a larger interval.

what is CSMA/CA?

reverse path forwarding is an approach to controlled flooding (avoids broadcast storm). When a router receives a broadcast packet with a given source address, it transmits the packet on all of its outgoing links (except the one on which it was received) only if the packet arrived on the link that is on its own shortest unicast path back to the source. Otherwise, the router simply discards the incoming packet without forwarding it on any of its outgoing links. Such a packet can be dropped because the router knows it either will receive or has already received a copy of this packet on the link that is on its own shortest path back to the sender. Figure 4.44 illustrates RPF. Suppose that the links drawn with thick lines represent the least-cost paths from the receivers to the source (A). Node A initially broadcasts a source-A packet to nodes C and B. Node B will forward the source-A packet it has received from A (since A is on its least-cost path to A) to both C and D. B will ignore (drop, without forwarding) any source-A packets it receives from any other nodes (for example, from routers C or D). Let us now consider node C, which will receive a source-A packet directly from A as well as from B. Since B is not on C's own shortest path back to A, C will ignore any source-A packets it receives from B. On the other hand, when C receives a source-A packet directly from A, it will forward the packet to nodes B, E, and F. (see page 402 for diagram)

what is RPF?

Let's now consider why hidden terminals can be problematic. Suppose Station H1 is transmitting a frame and halfway through H1's transmission, Station H2 wants to send a frame to the AP. H2, not hearing the transmission from H1, will first wait a DIFS interval and then transmit the frame, resulting in a collision. The channel will therefore be wasted during the entire period of H1's transmission as well as during H2's transmission. In order to avoid this problem, the IEEE 802.11 protocol allows a station to use a short Request to Send (RTS) control frame and a short Clear to Send (CTS) control frame to reserve access to the channel. When a sender wants to send a DATA frame, it can first send an RTS frame to the AP, indicating the total time required to transmit the DATA frame and the acknowledgment (ACK) frame. When the AP receives the RTS frame, it responds by broadcasting a CTS frame. This CTS frame serves two purposes: It gives the sender explicit permission to send and also instructs the other stations not to send for the reserved duration.

what is RTS/CTS?

basic service set (BSS). A BSS contains one or more wireless stations and a central base station, known as an access point (AP) in 802.11 parlance. Figure 6.7 shows the AP in each of two BSSs connecting to an interconnection device (such as a switch or router), which in turn leads to the Internet. In a typical home network, there is one AP and one router (typically integrated together as one unit) that connects the BSS to the Internet. http://i.imgur.com/aHjY10H.png

what is a BSS?

The base station is a key part of the wireless network infrastructure. Unlike the wireless host and wireless link, a base station has no obvious counterpart in a wired network. Abase station is responsible for sending and receiving data (e.g., packets) to and from a wireless host that is associated with that base station. A base station will often be responsible for coordinating the transmission of multiple wireless hosts with which it is associated. When we say a wireless host is "associated" with a base station, we mean that (1) the host is within the wireless communication distance of the base station, and (2) the host uses that base station to relay data between it (the host) and the larger network. Cell towers in cellular networks and access points in 802.11 wireless LANs are examples of base stations (wireless router or wireless access point).

what is a base station?

During flooding, When a node is connected to more than two other nodes, it will create and forward multiple copies of the broadcast packet, each of which will create multiple copies of itself (at other nodes with more than two neighbors), and so on. This broadcast storm, resulting from the endless multiplication of broadcast packets, would eventually result in so many broadcast packets being created that the network would be rendered useless. RPF and sequence-number-controlled-flooding are used to avoid broadcast storms.

what is a broadcast storm and what methods are used to avoid it?

When a mobile host moves beyond the range of one base station and into the range of another, it will change its point of attachment into the larger network (i.e., change the base station with which it is associated)—a process referred to as handoff.

what is a handoff?

As in the case of wired networks, hosts are the end-system devices that run applications. A wireless host might be a laptop, palmtop, smartphone, or desktop computer. The hosts themselves may or may not be mobile.

what is a wireless host?

A host connects to a base station or to another wireless host through a wireless communication link. Different wireless link technologies have different transmission rates and can transmit over different distances.

what is a wireless link?

The most obvious technique for achieving broadcast is a flooding approach in which the source node sends a copy of the packet to all of its neighbors. When a node receives a broadcast packet, it duplicates the packet and forwards it to all of its neighbors (except the neighbor from which it received the packet). Clearly, if the graph is connected, this scheme will eventually deliver a copy of the broadcast packet to all nodes in the graph. Although this scheme is simple and elegant, it has a fatal flaw: If the graph has cycles, then one or more copies of each broadcast packet will cycle indefinitely.

what is flooding?

Designed in late 1980s, widely deployed in the 1990s • Frame relay service: - No error control - End-to-end congestion control • Designed to interconnect corporate customer LANs - Typically permanent VCs: "pipe" carrying aggregate traffic between two routers - Switched VCs: as in ATM • Corporate customer leases FR service from public Frame Relay network (eg, Sprint, AT&T)

what is frame relay?

Hosts associated with a base station are often referred to as operating in infrastructure mode, since all traditional network services (e.g., address assignment and routing) are provided by the network to which a host is connected via the base station. In ad hoc networks, wireless hosts have no such infrastructure with which to connect. In the absence of such infrastructure, the hosts themselves must provide for services such as routing, address assignment, DNS-like name translation, and more.

what is infrastructure mode and ad-hoc mode?

A point-to-point link consists of a single sender at one end of the link and a single receiver at the other end of the link. Many link-layer protocols have been designed for point-to-point links; the point-to-point protocol (PPP) and high-level data link control (HDLC) are two such protocols that we'll cover later in this chapter. The second type of link, a broadcast link, can have multiple sending and receiving nodes all connected to the same, single, shared broadcast channel. The term broadcast is used here because when any one node transmits a frame, the channel broadcasts the frame and each of the other nodes receives a copy. Ethernet and wireless LANs are examples of broadcast link-layer technologies.

what is the difference between a point to point link and a broadcast link?

unicast is single source node to single destination node. broadcast is single source node to all other nodes in the network. multicast enables a single source node to send a copy of a packet to a subset of other network nodes. anycast routes datagrams to a single member of a group of potential receivers that are all identified by the same destination address. This is a one-to-nearest association.

what is the difference between unicast, broadcast, multicast, and anycast?

in the case of wired broadcast links, all nodes receive the transmissions from all other nodes. In the case of wireless links, the situation is not as simple. Suppose that Station A is transmitting to Station B. Suppose also that Station C is transmitting to Station B. With the so called hidden terminal problem, physical obstructions in the environment (for example, a mountain or a building) may prevent A and C from hearing each other's transmissions, even though A's and C's transmissions are indeed interfering at the destination, B. (This could cause collisions with A and C's transmissions). ------------------------------- A similar collision causing issue is fading where A and C are placed such that their signals are not strong enough to detect each other's transmissions, yet their signals are strong enough to interfere with each other at station B.

what is the hidden terminal problem?

X.25 is a standard suite of protocols used for packet switching across computer networks. The X.25 protocols works at the physical, data link, and network layers (Layers 1 to 3) of the OSI model.Each X.25 packets contains up to 128 bytes of data. The X.25 network handles packet assembly at the source device, delivery, and then dis-assembly at the destination. X.25 packet delivery technology includes not only switching and network-layer routing, but also error checking and re-transmission logic should delivery failures occur. X.25 supports multiple simultaneous conversations by multiplexing packets and using virtual communication channels.X.25 was originally designed more than 25 years ago to carry voice over analog telephone lines (dialup networks). Typical applications of X.25 today include automatic teller machine networks and credit card verification networks. X.25 also supports a variety of mainframe terminal/server applications.With the widespread acceptance of Internet Protocol (IP) as a standard for corporate networks, many X.25 applications are now being migrated to cheaper solutions using IP as the network layer protocol and replacing the lower layers of X.25 with Ethernet or ATM hardware.

what is x.25?

For the most part, the link layer is implemented in a network adapter, also sometimes known as a network interface card (NIC). At the heart of the network adapter is the link-layer controller, usually a single, special-purpose chip that implements many of the link-layer services (framing, link access, error detection, and so on). Thus, much of a link-layer controller's functionality is implemented in hardware.

what part does the network adapter play in the link layer?