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UNIT II
MEDIA ACCESS & INTERNETWORKING

Medium access control:

In telecommunications and computer networks, a channel access method or multiple

access method allows several terminals connected to the same multi-point transmission medium to
transmit over it and to share its capacity. Examples of shared physical media are wireless networks,
bus networks, ring networks and half-duplex point-to-point links.
A channel-access scheme is based on a multiplexing method, that allows several data
streams or signals to share the same communication channel or physical medium. Multiplexing is in
this context provided by the physical layer. Note that multiplexing also may be used in full-duplex
point-to-point communication between nodes in a switched network, which should not be
considered as multiple accesses.

A channel-access scheme is also based on a multiple access protocol and control mechanism,
also known as media access control (MAC). This protocol deals with issues such as addressing,
assigning multiplex channels to different users, and avoiding collisions. The MAC-layer is a sub-layer
in Layer 2 (Data Link Layer) of the OSI model and a component of the Link Layer of the TCP/IP
model.

Frequency Division Multiple Access (FDMA)

The frequency-division multiple access (FDMA) channel-access scheme is based on the

frequency-division multiplexing (FDM) scheme, which provides different frequency bands to
different data-streams. In the FDMA case, the data streams are allocated to different nodes or devices.
An example of FDMA systems were the first-generation (1G) cell-phone systems, where each phone
call was assigned to a specific uplink frequency channel, and another downlink frequency channel.
Each message signal (each phone call) is modulated on a specific carrier frequency.

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A related technique is wavelength division multiple access (WDMA), based on wavelength-

division multiplexing (WDM), where different datastreams get different colors in fiber-optical
communications. In the WDMA case, different network nodes in a bus or hub network get a different
color.

An advanced form of FDMA is the orthogonal frequency-division multiple access (OFDMA) scheme,
for example used in 4G cellular communication systems. In OFDMA, each node may use several sub-
carriers, making it possible to provide different quality of service (different data rates) to different
users. The assignment of sub-carriers to users may be changed dynamically, based on the current
radio channel conditions and traffic load.

Time division multiple access (TDMA)

The time division multiple access (TDMA) channel access scheme is based on the time-
division multiplexing (TDM) scheme, which provides different time-slots to different data-streams (in
the TDMA case to different transmitters) in a cyclically repetitive frame structure. For example, node
1 may use time slot 1, node 2 time slot 2, etc. until the last transmitter. Then it starts all over again, in
a repetitive pattern, until a connection is ended and that slot becomes free or assigned to another node.
An advanced form is Dynamic TDMA (DTDMA), where a scheduling may give different time
sometimes but some times node 1 may use time slot 1 in first frame and use another time slot in next
frame.

As an example, 2G cellular systems are based on a combination of TDMA and FDMA. Each
frequency channel is divided into eight timeslots, of which seven are used for seven phone calls, and
one for signaling data.

Code division multiple access (CDMA)/Spread spectrum multiple access (SSMA)

The code division multiple access (CDMA) scheme is based on spread spectrum, meaning that
a wider radio spectrum in Hertz is used than the data rate of each of the transferred bit streams, and
several message signals are transferred simultaneously over the same carrier frequency, utilizing
different spreading codes.

The wide bandwidth makes it possible to send with a very poor signal-to-noise ratio of much
less than 1 (less than 0 dB) according to the Shannon-Heartly formula, meaning that the transmission
power can be reduced to a level below the level of the noise and co-channel interference (cross talk)
from other message signals sharing the same frequency.

One form is direct sequence spread spectrum (DS-CDMA), used for example in 3G cell phone
systems. Each information bit (or each symbol) is represented by a long code sequence of several
pulses, called chips. The sequence is the spreading code, and each message signal (for example each
phone call) use different spreading code.

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Another form is frequency-hopping (FH-CDMA), where the channel frequency is changing

very rapidly according to a sequence that constitutes the spreading code. As an example, the
Bluetooth communication system is based on a combination of frequency-hopping and either
CSMA/CA packet mode communication (for data communication applications) or TDMA (for audio
transmission). All nodes belonging to the same user (to the same virtual private area network or
picante) use the same frequency hopping sequences synchronously, meaning that they send on the
same frequency channel, but CDMA/CA or TDMA is used to avoid collisions within the VPAN.
Frequency-hopping is used to reduce the cross-talk and collision probability between nodes in
different VPAN’s.

Ethernet:

Ethernet is a family of computer networking technologies for local area networks (LANs).
Ethernet was commercially introduced in 1980 and standardized in 1983 as IEEE 802.3.[1] Ethernet
has largely replaced competing wired LAN technologies such as token ring, FDDI, and ARCNET.

The Ethernet standards comprise several wiring and signaling variants of the OSI physical layer in use
with Ethernet. The original 10BASE5 Ethernet used coaxial cable as a shared medium. Later the
coaxial cables were replaced with twisted pair and fiber optic links in conjunction with hubs or
switches. Data rates were periodically increased from the original 10 megabits per second to 100
gigabits per second.

Systems communicating over Ethernet divide a stream of data into shorter pieces called
frames. Each frame contains source and destination addresses and error-checking data so that
damaged data can be detected and re-transmitted. As per the OSI model, Ethernet provides services
up to and including the data link layer.

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Since its commercial release, Ethernet has retained a good degree of compatibility. Features
such as the 48-bit MAC address and Ethernet frame format have influenced other networking
protocols.

Developed in the mid-1970s by researchers at the Xerox Palo Alto Research Center (PARC),
the Ethernet eventually became the dominant local area networking technology, emerging from a
pack of competing technologies. Today, it competes mainly with 802.11 wireless networks but
remains extremely popular in campus networks and data centers. The more general name for the
technology behind the Ethernet is Carrier Sense, Multiple Access with Collision Detect (CSMA/CD).
As indicated by the CSMA name, the Ethernet is a multiple-access network, meaning that a set of
nodes sends and receives frames over a shared link. You can, therefore, think of an Ethernet as being
like a bus that has multiple stations plugged into it. The “carrier sense” in CSMA/CD means that all
the nodes can distinguish between an idle and a busy link, and “collision detect” means that a node
listens as it transmits and can therefore detect when a frame it is transmitting has interfered (collided)
with a frame transmitted by another node.

The Ethernet has its roots in an early packet radio network, called Aloha, developed at the
University of Hawaii to support computer communication across the Hawaiian Islands. Like the
Aloha network, the fundamental problem faced by the Ethernet is how to mediate access to a shared
medium fairly and efficiently (in Aloha, the medium was the atmosphere, while in the Ethernet the
medium is a coax cable). The core idea in both Aloha and the Ethernet is an algorithm that controls
when each node can transmit. Interestingly, modern Ethernet links are now largely point to point; that
is, they connect one host to an Ethernet switch, or they interconnect switches. Hence, “multiple
access” techniques are not used much in today’s Ethernets. At the same time, wireless networks have
become enormously popular, so the multiple access technologies that started in Aloha are today again
mostly used in wireless networks such as 802.11 (Wi-Fi) networks.

We will discuss Ethernet switches in the next chapter. For now, we’ll focus on how a single
Ethernet link works. And even though multi-access Ethernet is becoming a bit of a historical
curiosity, the principles of multi-access networks continue to be important enough to warrant some
further discussion, which we provide below. Digital Equipment Corporation and Intel Corporation
joined Xerox to define a 10-Mbps Ethernet standard in 1978. This standard then formed the basis for
IEEE standard 802.3, which additionally defines a much wider collection of physical media over
which an Ethernet can operate, including 100-Mbps, 1-Gbps, and 10-Gbps versions.

Physical Properties:

Ethernet segments were originally implemented using coaxial cable of length up to 500 m.
(Modern Ethernets use twisted copper pairs, usually a particular type known as “Category 5,” or
optical fibers, and in some cases can be quite a lot longer than 500 m.) This cable was similar to the
type used for cable TV. Hosts connected to an Ethernet segment by tapping into it. A transceiver, a
small device directly attached to the tap, detected when the line was idle and drove the signal when

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the host was transmitting. It also received incoming signals. The transceiver, in turn, connected to an
Ethernet adaptor, which was plugged into the host.

Multiple Ethernet segments can be joined together by repeaters. A repeater is a device that
forwards digital signals, much like an amplifier Transceiver Ethernet cable Adaptor Host.

Wireless LAN’s:

Wireless technologies differ from wired links in some important ways, while at the same time
sharing many common properties. Like wired links, issues of bit errors are of great concern—
typically even more so due to the predictable noise environment of most wireless links. Framing and
reliability also have to be addressed. Unlike wired links, power is a big issue for wireless, especially
because wireless links are often used by small mobile devices (like phones and sensors) that have
limited access to power (e.g., a small battery). Furthermore, you can’t go blasting away at arbitrarily
high power with a radio transmitter—there are concerns about interference with other devices and
usually regulations about how much power a device may emit at any given frequency. Wireless media
are also inherently multi-access; it’s difficult to direct your radio transmission to just a single receiver
or to avoid receiving radio signals from any transmitter with enough power in your neighborhood.
Hence, media access control is a central issue for wireless links. And, because it’s hard to control who
receives your signal when you transmit over the air, issues of eavesdropping may also have to be
addressed.

Wi-Fi (more formally known as 802.11), Bluetooth, and the third-generation or “3G” family of
cellular wireless standards. Frequency band in hertz and sometimes the data rate of a link. Because
both these concepts come up in discussions of wireless networks, we’re going to use bandwidth here
in its stricter sense—width of a frequency band—and use the term data rate to describe the number of
bits per second that can be sent over the link. Because wireless links all share the same medium, the
challenge is to share that medium efficiently, without unduly interfering with each other. Most of this
sharing is accomplished by dividing it up along the dimensions of frequency and space. Exclusive use
of a particular frequency in a particular geographic area may be allocated to an individual entity such
as a corporation. It is feasible to limit the area covered by an electromagnetic signal because such
signals weaken, or attenuate, with the distance from their origin. To reduce the area covered by your
signal, reduce the power of your transmitter. These allocations are typically determined by
government agencies, such as the Federal Communications Commission (FCC) in the United States.
Specific bands (frequency ranges) are allocated to certain uses. Some bands are reserved for

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government use. Other bands are reserved for uses such as AM radio, FM radio, television, satellite
communication, and cellular phones.

A second spread spectrum technique, called direct sequence, adds redundancy for greater
tolerance of interference. Each bit of data is represented by multiple bits in the transmitted signal so
that, if some of the transmitted bits are damaged by interference, there is usually enough redundancy
to recover the original bit. For each bit the sender wants to transmit, it actually sends the exclusive-
OR of that bit and n random bits. As with frequency hopping, the sequence of random bits is
generated by a pseudorandom number generator known to both the sender and the receiver. The
transmitted values, known as an n-bit chipping code, spread the signal across a frequency band that is
n times wider than the frame would have otherwise required.

802.11/Wi-Fi
Most readers will have used a wireless network based on the IEEE 802.11standards, often
referred to as Wi-Fi.9 Wi-Fi is technically a trademark, owned by a trade group called the Wi-Fi
Alliance, which certifies product compliance with 802.11. Like Ethernet, 802.11 is designed for use in
a limited geographical area (homes, office buildings, campuses), and its primary challenge is to
mediate access to a shared communication medium—in this case, signals propagating through space.
Physical Properties 802.11 defines a number of different physical layers that operate in various
frequency bands and provide a range of different data rates. At the time of writing, 802.11n provides
the highest maximum data rate, topping out at 600 Mbps. The original 802.11 standard defined two
radio-based physical layers standards, one using frequency hopping (over 79 1-MHz-wide frequency
bandwidths) and the other using direct sequence spread spectrum (with an 11-bit chipping sequence).
Both provided data rates in the 2 Mbps range. The physical layer standard 802.11b was added
subsequently. Using a variant of direct sequence, 802.11b provides up to 11 Mbps. These three
standards all operated in the license-exempt 2.4-GHz frequency band of the electromagnetic
spectrum. Then came 802.11a, which delivers up to 54 Mbps using a variant of FDM called
orthogonal frequency division multiplexing (OFDM); 802.11a runs in the license-exempt 5-GHz
band. On one hand, this band is less used, so there is less interference. On the other hand, there is
more absorption of the signal and it is limited to almost line of sight. 802.11g followed; 802.11g also
uses OFDM, delivers up to 54 Mbps, and is backward compatible with 802.11b (and returns to the
2.4-GHz band). Most recently 802.11n has appeared on the scene, with a standard that was approved
in 2009 (although pre-standard products also existed). 802.11n achieves considerable advances in
maximum possible data rate using multiple antennas and allowing greater wireless channel
bandwidths. The use of multiple antennas is often called MIMO for multiple input, multiple-output.

The systems try to pick an optimal bit rate based on the noise environment in which they
find themselves; the algorithms for bit rate selection can be quite complex (see the Further Reading
section for an example). Interestingly, the 802.11 standards do not specify a particular approach but
leave the algorithms to the various vendors. The basic approach to picking a bit rate is to estimate the
bit error rate either by directly measuring the signal-to-noise ratio (SNR) at the physical layer or by
estimating the SNR by measuring how often packets are successfully transmitted and acknowledged.
In some approaches, a sender will occasionally probe a higher bit rate by sending one or more
packets at that rate to see if it succeeds.

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Distribution System:

As described so far, 802.11 would be suitable for a network with a mesh (ad hoc)
topology, and development of a 802.11s standard for mesh networks is nearing completion. At
the current time, however, nearly all 802.11 networks use a base-station-oriented topology.
Instead of all nodes being created equal, some nodes are allowed to roam (e.g., your laptop) and
some are connected to a wired network infrastructure. 802.11 calls these base stations access
points (APs), and they are connected to each other by a so-called distribution system.

A distribution system that connects three access points, each of which services the nodes
in some region. Each access point operates on some channel in the appropriate frequency range,
and each AP will typically be on a different channel than its neighbors. The details of the
distribution system are not important to this discussion—it could be an Ethernet, for example.
The only important point is that the distribution network operates at the link layer, the same
protocol layer as the wireless links. In other words, it does not depend on any higher-level
protocols (such as the network layer).

Bluetooth:

Bluetooth fills the niche of very short range communication between mobile phones,
PDAs, notebook computers, and other personal or peripheral devices. For example, Bluetooth
can be used to connect a mobile phone to a headset or a notebook computer to a keyboard.
Roughly speaking, Bluetooth is a more convenient alternative to connecting two devices with a
wire. In such applications, it is not necessary to provide much range or bandwidth. This means
that Bluetooth radios can use quite low power transmission, since transmission power is one of
the main factors affecting bandwidth and range of wireless links. This matches the target
applications for Bluetooth-enabled devices—most of them are battery powered (such as the
ubiquitous phone headset) and hence it is important that they not consume much power.

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Bluetooth operates in the license-exempt band at 2.45 GHz. Bluetooth links have typical
bandwidths around 1 to 3 Mbps and a range of about 10 m. For this reason, and because the
communicating devices typically belong to one individual or group, Bluetooth is sometimes
categorized as a Personal Area Network (PAN).

Bluetooth is specified by an industry consortium called the Bluetooth Special Interest Group.
It specifies an entire suite of protocols, going beyond the link layer to define application protocols,
which it calls profiles, for a range of applications. For example, there is a profile for synchronizing a
PDA with a personal computer. Another profile gives a mobile computer access to a wired LAN in
the manner of 802.11, although this was not Bluetooth’s original goal. The IEEE 802.15.1 standard is
based on Bluetooth but excludes the application protocols.

The basic Bluetooth network configuration, called a picante, consists of a master device and
up to seven slave devices. Any communication is between the master and a slave; the slaves do not
communicate directly with each other. Because slaves have a simpler role, their Bluetooth hardware
and software can be simpler and cheaper. Since Bluetooth operates in an license-exempt band, it is
required to use a spread spectrum technique. To deal with possible interference in the band. It uses
frequency-hopping with 79 channels (frequencies), using each for 625 μs at a time. This provides a
natural time slot for Bluetooth to use for synchronous time division multiplexing. A frame takes up 1,
3, or 5 consecutive time slots. Only the master can start to transmit in odd-numbered slots. A slave
can start to transmit in an even-numbered slot—but only in response to a request from the master
during the previous slot, thereby preventing any contention between the slave devices. A slave device
can be parked; that is, it is set to an inactive, low-power state. A parked device cannot communicate
on the picante; it can only be reactivated by the master. A picante can have up to 255 parked devices
in addition to its active slave devices.
Switching And Bridging:

In the simplest terms, a switch is a mechanism that allows us to interconnect links to

form a larger network. A switch is a multi-input, multi-output device that transfers packets from
an input to one or more outputs. Thus, a switch adds the star topology to the point-to-point link,
bus (Ethernet), and ring topologies established in the last chapter.

A star topology has several attractive properties: Even though a switch has a fixed
number of inputs and outputs, which limits the number of hosts that can be connected to a single
switch, large networks can be built by interconnecting a number of switches.

We can connect switches to each other and to hosts using point-to-point links, which
typically means that we can build networks of large geographic scope. n Adding a new host to
the network by connecting it to a switch does not necessarily reduce the performance of the
network for other hosts already connected. This last claim cannot be made for the shared-media
networks.

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For example, it is impossible for two hosts on the same 10-Mbps Ethernet segment to
transmit continuously at 10 Mbps because they share the same transmission medium. Every host on a
switched network has its own link to the switch, so it may be entirely possible for many hosts to
transmit at the full link speed (bandwidth), provided that the switch is designed with enough
aggregate capacity. Providing high aggregate throughput is one of the design goals for a switch; we
return to this topic later. In general, switched networks are considered more scalable (i.e., more
capable of growing to large numbers of nodes) than shared-media networks because of this ability to
support many hosts at full speed. A switch is connected to a set of links and, for each of these links,
runs the appropriate data link protocol to communicate with the node at the other end of the link. A
switch’s primary job is to receive incoming packets on one of its links and to transmit them on some
other link.

This function is sometimes referred to as either switching or forwarding, and in terms of the
Open Systems Interconnection (OSI) architecture, it is the main function of the network layer. The
first is the datagram or connectionless approach. The second is the virtual circuit or connection-
oriented approach. A third approach, source routing, is less common than these other two, but it does
have some useful applications. One thing that is common to all networks is that we need to have a
way to identify the end nodes. Such identifiers are usually called addresses.

We have already seen examples of addresses in the previous chapter, such as the 48-bit
address used for Ethernet. The only requirement for Ethernet addresses is that no two nodes on a
network have the same address. This is accomplished by making sure that all Ethernet cards are
assigned a globally unique identifier. For the following discussions, we assume that each host has a
globally unique address. Later on, we consider other useful properties that an address might have, but
global uniqueness is adequate to get us started. Another assumption that we need to make is that there
is some way to identify the input and output ports of each switch. There are at least two sensible ways
to identify ports: One is to number each port, and the other is to identify the port by the name of the

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node (switch or host) to which it leads. For now, we use numbering of the ports.

Datagrams:
The idea behind datagrams is incredibly simple: You just include in every packet enough information
to enable any switch to decide how to get it to its destination. That is, every packet contains the
complete destination address. Consider the example network in which the hosts have addresses A, B,
C, and so on. To decide how to forward a packet, a switch consults a forwarding table (sometimes
called a routing table). This particular table shows the forwarding information that switch 2 needs to
forward datagrams in the example network. It is pretty easy to figure out such a table when you have
a complete map of a simple network like that depicted here; we could imagine a network operator
configuring Datagram networks have the following characteristics: n A host can send a packet
anywhere at any time, since any packet that turns up at a switch can be immediately forwarded
(assuming a correctly populated forwarding table). For this reason, datagram networks are often
called connectionless; this contrasts with the connection-oriented networks described below, in which
some connection state needs to be established before the first data packet is sent. n When a host sends
a packet, it has no way of knowing if the network is capable of delivering it or if the destination host
is even up and running. n Each packet is forwarded independently of previous packets that might have
been sent to the same destination. Thus, two successive packets from host A to host B may follow
completely different paths (perhaps because of a change in the forwarding table at some switch in the
network).
A switch or link failure might not have any serious effect on communication if it is
possible to find an alternate route around the failure and to update the forwarding table accordingly.
This last fact is particularly important to the history of datagram networks. One of the important
design goals of the Internet is robustness to failures.

Virtual Circuit Switching:

A second technique for packet switching, which differs significantly from the datagram
model, uses the concept of a virtual circuit (VC). A virtual circuit identifier (VCI) hat uniquely
identifies the connection at this switch and which will be carried inside the header of the packets that
belong to this connection n An incoming interface on which packets for this VC arrive at the switch
n An outgoing interface in which packets for this VC leave the switch. A potentially different VCI
that will be used for outgoing packets The semantics of one such entry is as follows: If a packet
arrives on the designated incoming interface and that packet contains the designated VCI value in its
header, then that packet should be sent out the specified outgoing interface with the specified
outgoing VCI value having been first placed in its header.

Basic Internetworking (IP, CIDR, ARP, DHCP,ICMP ):

BASIC INTERNETWORKING (IP):
In the previous section, we saw that it was possible to build reasonably large LANs
using bridges and LAN switches, but that such approaches were limited in their ability to scale
and to handle heterogeneity. In this section, we explore some ways to go beyond the limitations
of bridged networks, enabling us to build large, highly heterogeneous networks with reasonably
efficient routing. We refer to such networks as internetworks. We’ll continue the discussion of

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how to build a truly global internetworking the next chapter, but for now we’ll explore the
basics. We start by considering more carefully what the word internetwork means.

What Is an Internetwork?

We use the term internetwork, or sometimes just internet with a lowercase i, to refer to an
arbitrary collection of networks interconnected to provide some sort of host-to-host packet delivery
service. For example, a corporation with many sites might construct a private internetwork by
interconnecting the LANs at their different sites with point-to-point links leased from the phone
company. When we are talking about the widely used global internetwork to which a large percentage
of networks are now connected, we call it the Internet with a capital I. In keeping with the first
principles approach of this book, we mainly want you to learn about the principles of “lowercase i”
internetworking, but we illustrate these ideas with real-world examples from the “big I” Internet.
Another piece of terminology that can be confusing is the difference between networks, sub-networks,
and internetworks. We are going to avoid sub-networks (or subnets) altogether. For now, we use
network to mean either a directly connected or a switched network.

An inter-network is an interconnected collection of such networks. Sometimes, to avoid

ambiguity, we refer to the underlying networks that we are interconnecting as physical networks. An
internet is a logical network built out of a collection of physical networks. In this context, a collection
of Ethernets connected by bridges or switches would still be viewed as a single network. An
internetwork is often referred to as a “network of networks” because it is made up of lots of smaller
networks. In this figure, we see Ethernets, a wireless network, The Internet Protocol is the key tool
used today to build scalable, heterogeneous internetworks. It was originally known as the Kahn-Cerf
protocol after its inventors.6 One way to think of IP is that it runs on all the nodes (both hosts and
routers) in a collection of networks and defines the infrastructure that allows these nodes and
networks to function as a single logical internetwork. Such as TCP and UDP, typically run on top of
IP on the hosts.

Service Model

A good place to start when you build an internetwork is to define its service model, that is, the

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host-to-host services you want to provide. The main concern in defining a service model for an
internetwork is that we can provide a host-to-host service only if this service can somehow be
provided over each of the underlying physical networks. For example, it would be no good deciding
that our internetwork service model was going to provide guaranteed delivery of every packet in 1 ms
or less if there were underlying network technologies that could arbitrarily delay packets. The
philosophy used in defining the IP service model, therefore, was to make it undemanding enough that
just about any network technology that might turn up in an internetwork would be able to provide the
necessary service. The IP service model can be thought of as having two parts: an addressing scheme,
which provides a way to identify all hosts in the internetwork, and a datagram (connectionless) model
of data delivery. This service model is sometimes called best effort because, although IP makes every
effort to deliver datagrams, it makes no guarantees. We postpone a discussion of the addressing
scheme for now and look first at the data delivery model.

Datagram Delivery

The IP datagram is fundamental to the Internet Protocol. A datagram is a type of packet that
happens to be sent in a connectionless manner over a network. Every datagram carries enough
information to let the network forward the packet to its correct destination; there is no need for any
advance setup mechanism to tell the network what to do when the packet arrives. You just send it, and
the network makes its best effort to get it to the desired destination. The “best-effort” part means that
if something goes wrong and the packet gets lost, corrupted, miss delivered, or in any way fails to
reach its intended destination, the network does nothing—it made its best effort, and that is all it has
to do. It does not make any attempt to recover from the failure. This is sometimes called an unreliable
service.

Clearly, a key part of the IP service model is the type of packets that can be carried. The IP
datagram, like most packets, consists of a header followed by a number of bytes of data. A different
style of representing packets than the one we used in previous chapters. This is because packet
formats at the internetworking layer and above, where we will be focusing our attention.

CIDR:

CIDR, therefore, tries to balance the desire to minimize the number of routes that a router
needs to know against the need to hand out addresses efficiently. To do this, CIDR helps us to
aggregate routes. That is, it lets us use a single entry in a forwarding table to tell us how to reach a lot
of different networks. As noted above it does this by breaking the rigid boundaries between address
classes. To understand how this works, consider our hypothetical organization with 16 class C
network numbers.

Instead of handing out 16 addresses at random, we can hand out a block of contiguous class
C addresses. Suppose we assign the class C network numbers from 192.4.16 through 192.4.31.
Observe that the top 20 bits of all the addresses in this range are the same (11000000 00000100
0001). Thus, what we have effectively created is a 20-bit network number—something that is
between a class B network number and a class C number in terms of the number of hosts that it can
support. In other words, we get both the high address efficiency of handing out addresses in chunks
smaller than a class B network, and a single network prefix that can be used in forwarding tables.
Observe that, for this scheme to work, we need to hand out blocks of class C addresses that share a

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common prefix, which means that each block must contain a number of class C networks that is a
power of two.

CIDR requires a new type of notation to represent network numbers, or prefixes as they are
known, because the prefixes can be of any length. The convention is to place a /X after the prefix,
where X is the prefix length in bits. So, for the example above, the 20-bit prefix for all the networks
192.4.16 through 192.4.31 is represented as 192.4.16/20. By contrast, if we wanted to represent a
single class C network number, which is 24 bits long, we would write it 192.4.16/24. Today, with
CIDR being the norm, it is more common to hear people talk about “slash 24” prefixes than class C
networks. Note that representing a network address in this way is similar to the h mask, value i
approach used in sub netting, as long as masks consist of contiguous bits starting from the most
significant bit (which in practice is almost always the case).

Address Translation (ARP):

In the previous section we talked about how to get IP datagrams to the right physical network
but glossed over the issue of how to get a datagram to a particular host or router on that network. The
main issue is that IP datagrams contain IP addresses, but the physical interface hardware on the host
or router to which you want to send the datagram only understands the addressing scheme of that
particular network. Thus, we need to translate the IP address to a link-level address that makes sense
on this network (e.g., a 48-bit Ethernet address). We can then encapsulate the IP datagram inside a
frame that contains that link-level address and send it either to the ultimate destination or to a router
that promises to forward the datagram toward the ultimate destination. One simple way to map an IP
address into a physical network address is to encode a host’s physical address in the host part of its IP
address.

For example, a host with physical address 00100001 01001001 (which has the decimal
value 33 in the upper byte and 81 in the lower byte) might be given the IP address 128.96.33.81.
While this solution has been used on some networks, it is limited in that the network’s physical
addresses can be no more than 16 bits long in this example; they can be only 8 bits long on a class C
network. This clearly will not work for 48-bit Ethernet addresses.

Host Configuration (DHCP):

Ethernet addresses are configured into the network adaptor by the manufacturer, and this
process is managed in such a way to ensure that these addresses are globally unique. This is clearly a
sufficient condition to ensure that any collection of hosts connected to a single Ethernet (including an
extended LAN) will have unique addresses. Furthermore, uniqueness is all we ask of Ethernet
addresses. IP addresses, by contrast, not only must be unique on a given internetwork but also must
reflect the structure of the internetwork. As noted above, they contain a network part and a host part,
and the network part must be the same for all hosts on the same network.

Thus, it is not possible for the IP address to be configured once into a host when it is
manufactured, since that would imply that the manufacturer knew which hosts were going to end up
on which networks, and it would mean that a host, once connected to one network, could never move
to another. For this reason, IP addresses need to be reconfigurable. In addition to an IP address, there

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Course Material –Lecture Notes

are some other pieces of information a host needs to have before it can start sending packets. The
most notable of these is the address of a default router—the place to which it can send packets whose
destination address is not on the same network as the sending host.

Error Reporting (ICMP):

The next issue is how the Internet treats errors. While IP is perfectly willing to drop
datagrams when the going gets tough—for example, when a router does not know how to forward the
datagram or when one fragment of a datagram fails to arrive at the destination—it does not
necessarily fail silently. IP is always configured with a companion protocol, known as the Internet
Control Message Protocol (ICMP), that defines a collection of error messages that are sent back to
the source host whenever a router or host is unable to process an IP datagram successfully.

For example, ICMP defines error messages indicating that the destination host is
unreachable (perhaps due to a link failure), that the reassembly process failed, that the TTL had
reached 0 and that the IP header checksum failed, and so on. ICMP also defines a handful of control
messages that a router can send back to a source host. One of the most useful control messages, called
an ICMP-Redirect, tells the source host that there is a better route to the destination.

ICMP-Redirects are used in the following situation. Suppose a host is connected to a

network that has two routers attached to it, called R1 and R2, where the host uses R1 as its default
router. Should R1 ever receive a datagram from the host, where based on its forwarding table it knows
that R2 would have been a better choice for a particular destination address, it sends an ICMP-
Redirect back to the host, instructing it to use R2 for all future datagrams addressed to that
destination. The host then adds this new route to its forwarding table.

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