3.2 Basic Internetworking
3.2 Basic Internetworking
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 how to build a truly global internetwork in 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, subnetworks, and internetworks. We are going to avoid subnetworks (or subnets) altogether until a later section. For now, we use network to mean either a directly connected or a switched network of the kind described in the previous section and the previous chapter. Such a network uses one technology, such as 802.11 or Ethernet. An internetwork 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 Ethernet segments connected by bridges or switches would still be viewed as a single network.
Figure 1 shows an example internetwork. 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, and a point-to-point link. Each of these is a single-technology network. The nodes that interconnect the networks are called routers. They are also sometimes called gateways, but since this term has several other connotations, we restrict our usage to router.
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. 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. For example, Figure 2 shows how hosts H5 and H8 are logically connected by the internet in Figure 1, including the protocol graph running on each node. Note that higher-level protocols, such as TCP and UDP, typically run on top of IP on the hosts.
The rest of this and the next chapter are about various aspects of IP. While it is certainly possible to build an internetwork that does not use IP—and in fact, in the early days of the Internet there were alternative solutions—IP is the most interesting case to study simply because of the size of the Internet. Said another way, it is only the IP Internet that has really faced the issue of scale. Thus, it provides the best case study of a scalable internetworking protocol.
A good place to start when you build an internetwork is to define its service model, that is, the 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.
The IP datagram is fundamental to the Internet Protocol. Recall an earlier section that 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, misdelivered, 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.
Best-effort, connectionless service is about the simplest service you could ask for from an internetwork, and this is a great strength. For example, if you provide best-effort service over a network that provides a reliable service, then that's fine—you end up with a best-effort service that just happens to always deliver the packets. If, on the other hand, you had a reliable service model over an unreliable network, you would have to put lots of extra functionality into the routers to make up for the deficiencies of the underlying network. Keeping the routers as simple as possible was one of the original design goals of IP.
The ability of IP to "run over anything" is frequently cited as one of its most important characteristics. It is noteworthy that many of the technologies over which IP runs today did not exist when IP was invented. So far, no networking technology has been invented that has proven too bizarre for IP; in principle, IP can run over a network that transports messages using carrier pigeons.
Best-effort delivery does not just mean that packets can get lost. Sometimes they can get delivered out of order, and sometimes the same packet can get delivered more than once. The higher-level protocols or applications that run above IP need to be aware of all these possible failure modes.
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. The format of the header is shown in Figure 3. Note that we have adopted 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 for the next few chapters, are almost invariably designed to align on 32-bit boundaries to simplify the task of processing them in software. Thus, the common way of representing them (used in Internet Requests for Comments, for example) is to draw them as a succession of 32-bit words. The top word is the one transmitted first, and the leftmost byte of each word is the one transmitted first. In this representation, you can easily recognize fields that are a multiple of 8 bits long. On the odd occasion when fields are not an even multiple of 8 bits, you can determine the field lengths by looking at the bit positions marked at the top of the packet.
Looking at each field in the IP header, we see that the "simple" model
of best-effort datagram delivery still has some subtle features. The
Version field specifies the version of IP. The still-assumed version
of IP is 4, which is typically called IPv4. Observe that putting this
field right at the start of the datagram makes it easy for everything
else in the packet format to be redefined in subsequent versions; the
header processing software starts off by looking at the version and then
branches off to process the rest of the packet according to the
appropriate format. The next field,
HLen, specifies the length of the
header in 32-bit words. When there are no options, which is most of the
time, the header is 5 words (20 bytes) long. The 8-bit
TOS (type of
service) field has had a number of different definitions over the years,
but its basic function is to allow packets to be treated differently
based on application needs. For example, the
TOS value might determine
whether or not a packet should be placed in a special queue that
receives low delay.
The next 16 bits of the header contain the
Length of the datagram,
including the header. Unlike the
HLen field, the
Length field counts
bytes rather than words. Thus, the maximum size of an IP datagram is
65,535 bytes. The physical network over which IP is running, however,
may not support such long packets. For this reason, IP supports a
fragmentation and reassembly process. The second word of the header
contains information about fragmentation, and the details of its use are
presented in the following section entitled "Fragmentation and
Moving on to the third word of the header, the next byte is the
(time to live) field. Its name reflects its historical meaning rather
than the way it is commonly used today. The intent of the field is to
catch packets that have been going around in routing loops and discard
them, rather than let them consume resources indefinitely. Originally,
TTL was set to a specific number of seconds that the packet would be
allowed to live, and routers along the path would decrement this field
until it reached 0. However, since it was rare for a packet to sit for
as long as 1 second in a router, and routers did not all have access to
a common clock, most routers just decremented the
TTL by 1 as they
forwarded the packet. Thus, it became more of a hop count than a timer,
which is still a perfectly good way to catch packets that are stuck in
routing loops. One subtlety is in the initial setting of this field by
the sending host: Set it too high and packets could circulate rather a
lot before getting dropped; set it too low and they may not reach their
destination. The value 64 is the current default.
Protocol field is simply a demultiplexing key that identifies the
higher-level protocol to which this IP packet should be passed. There
are values defined for the TCP (Transmission Control Protocol—6), UDP
(User Datagram Protocol—17), and many other protocols that may sit
above IP in the protocol graph.
Checksum is calculated by considering the entire IP header as a
sequence of 16-bit words, adding them up using ones' complement
arithmetic, and taking the ones' complement of the result. Thus, if any
bit in the header is corrupted in transit, the checksum will not contain
the correct value upon receipt of the packet. Since a corrupted header
may contain an error in the destination address—and, as a result, may
have been misdelivered—it makes sense to discard any packet that fails
the checksum. It should be noted that this type of checksum does not
have the same strong error detection properties as a CRC, but it is much
easier to calculate in software.
The last two required fields in the header are the
SourceAddr and the
DestinationAddr for the packet. The latter is the key to datagram
delivery: Every packet contains a full address for its intended
destination so that forwarding decisions can be made at each router. The
source address is required to allow recipients to decide if they want to
accept the packet and to enable them to reply. IP addresses are
discussed in a later section—for now, the important thing
to know is that IP defines its own global address space, independent of
whatever physical networks it runs over. As we will see, this is one of
the keys to supporting heterogeneity.
Finally, there may be a number of options at the end of the header. The
presence or absence of options may be determined by examining the header
HLen) field. While options are used fairly rarely, a complete
IP implementation must handle them all.
Fragmentation and Reassembly
One of the problems of providing a uniform host-to-host service model over a heterogeneous collection of networks is that each network technology tends to have its own idea of how large a packet can be. For example, classic Ethernet can accept packets up to 1500 bytes long, but modern-day variants can deliver larger (jumbo) packets that carry up to 9000 bytes of payload. This leaves two choices for the IP service model: Make sure that all IP datagrams are small enough to fit inside one packet on any network technology, or provide a means by which packets can be fragmented and reassembled when they are too big to go over a given network technology. The latter turns out to be a good choice, especially when you consider the fact that new network technologies are always turning up, and IP needs to run over all of them; this would make it hard to pick a suitably small bound on datagram size. This also means that a host will not send needlessly small packets, which wastes bandwidth and consumes processing resources by requiring more headers per byte of data sent.
The central idea here is that every network type has a maximum transmission unit (MTU), which is the largest IP datagram that it can carry in a frame. Note that this value is smaller than the largest packet size on that network because the IP datagram needs to fit in the payload of the link-layer frame.
In ATM networks, the MTU is, fortunately, much larger than a single cell, as ATM has its own fragmentation mechanisms. The link-layer frame in ATM is called a convergence-sublayer protocol data unit (CS-PDU).
When a host sends an IP datagram, therefore, it can choose any size that it wants. A reasonable choice is the MTU of the network to which the host is directly attached. Then, fragmentation will only be necessary if the path to the destination includes a network with a smaller MTU. Should the transport protocol that sits on top of IP give IP a packet larger than the local MTU, however, then the source host must fragment it.
Fragmentation typically occurs in a router when it receives a datagram
that it wants to forward over a network that has an MTU that is smaller
than the received datagram. To enable these fragments to be reassembled
at the receiving host, they all carry the same identifier in the
field. This identifier is chosen by the sending host and is intended to
be unique among all the datagrams that might arrive at the destination
from this source over some reasonable time period. Since all fragments
of the original datagram contain this identifier, the reassembling host
will be able to recognize those fragments that go together. Should all
the fragments not arrive at the receiving host, the host gives up on the
reassembly process and discards the fragments that did arrive. IP does
not attempt to recover from missing fragments.
To see what this all means, consider what happens when host H5 sends a datagram to host H8 in the example internet shown in Figure 1. Assuming that the MTU is 1500 bytes for the two Ethernets and the 802.11 network, and 532 bytes for the point-to-point network, then a 1420-byte datagram (20-byte IP header plus 1400 bytes of data) sent from H5 makes it across the 802.11 network and the first Ethernet without fragmentation but must be fragmented into three datagrams at router R2. These three fragments are then forwarded by router R3 across the second Ethernet to the destination host. This situation is illustrated in Figure 4. This figure also serves to reinforce two important points:
Each fragment is itself a self-contained IP datagram that is transmitted over a sequence of physical networks, independent of the other fragments.
Each IP datagram is re-encapsulated for each physical network over which it travels.
The fragmentation process can be understood in detail by looking at the
header fields of each datagram, as is done in Figure 5.
The unfragmented packet, shown at the top, has 1400 bytes of data and a
20-byte IP header. When the packet arrives at router R2, which has an
MTU of 532 bytes, it has to be fragmented. A 532-byte MTU leaves
512 bytes for data after the 20-byte IP header, so the first fragment
contains 512 bytes of data. The router sets the M bit in the
field (see Figure 3), meaning that there are more
fragments to follow, and it sets the
Offset to 0, since this
fragment contains the first part of the original datagram. The data
carried in the second fragment starts with the 513th byte of the
original data, so the
Offset field in this header is set to 64,
which is 512/8. Why the division by 8? Because the designers of IP
decided that fragmentation should always happen on 8-byte boundaries,
which means that the
Offset field counts 8-byte chunks, not
bytes. (We leave it as an exercise for you to figure out why this
design decision was made.) The third fragment contains the last
376 bytes of data, and the offset is now 2 512/8
= 128. Since this is the last fragment, the M bit is not set.
Observe that the fragmentation process is done in such a way that it could be repeated if a fragment arrived at another network with an even smaller MTU. Fragmentation produces smaller, valid IP datagrams that can be readily reassembled into the original datagram upon receipt, independent of the order of their arrival. Reassembly is done at the receiving host and not at each router.
IP reassembly is far from a simple process. For example, if a single fragment is lost, the receiver will still attempt to reassemble the datagram, and it will eventually give up and have to garbage-collect the resources that were used to perform the failed reassembly. Getting a host to tie up resources needlessly can be the basis of a denial-of-service attack.
For this reason, among others, IP fragmentation is generally considered a good thing to avoid. Hosts are now strongly encouraged to perform "path MTU discovery," a process by which fragmentation is avoided by sending packets that are small enough to traverse the link with the smallest MTU in the path from sender to receiver.
In the above discussion of the IP service model, we mentioned that one of the things that it provides is an addressing scheme. After all, if you want to be able to send data to any host on any network, there needs to be a way of identifying all the hosts. Thus, we need a global addressing scheme—one in which no two hosts have the same address. Global uniqueness is the first property that should be provided in an addressing scheme.
Ethernet addresses are globally unique, but that alone does not suffice for an addressing scheme in a large internetwork. Ethernet addresses are also flat, which means that they have no structure and provide very few clues to routing protocols. (In fact, Ethernet addresses do have a structure for the purposes of assignment—the first 24 bits identify the manufacturer—but this provides no useful information to routing protocols since this structure has nothing to do with network topology.) In contrast, IP addresses are hierarchical, by which we mean that they are made up of several parts that correspond to some sort of hierarchy in the internetwork. Specifically, IP addresses consist of two parts, usually referred to as a network part and a host part. This is a fairly logical structure for an internetwork, which is made up of many interconnected networks. The network part of an IP address identifies the network to which the host is attached; all hosts attached to the same network have the same network part in their IP address. The host part then identifies each host uniquely on that particular network. Thus, in the simple internetwork of Figure 1, the addresses of the hosts on network 1, for example, would all have the same network part and different host parts.
Note that the routers in Figure 1 are attached to two networks. They need to have an address on each network, one for each interface. For example, router R1, which sits between the wireless network and an Ethernet, has an IP address on the interface to the wireless network whose network part is the same as all the hosts on that network. It also has an IP address on the interface to the Ethernet that has the same network part as the hosts on that Ethernet. Thus, bearing in mind that a router might be implemented as a host with two network interfaces, it is more precise to think of IP addresses as belonging to interfaces than to hosts.
Now, what do these hierarchical addresses look like? Unlike some other forms of hierarchical address, the sizes of the two parts are not the same for all addresses. Originally, IP addresses were divided into three different classes, as shown in Figure 6, each of which defines different-sized network and host parts. (There are also class D addresses that specify a multicast group and class E addresses that are currently unused.) In all cases, the address is 32 bits long.
The class of an IP address is identified in the most significant few bits. If the first bit is 0, it is a class A address. If the first bit is 1 and the second is 0, it is a class B address. If the first two bits are 1 and the third is 0, it is a class C address. Thus, of the approximately 4 billion possible IP addresses, half are class A, one-quarter are class B, and one-eighth are class C. Each class allocates a certain number of bits for the network part of the address and the rest for the host part. Class A networks have 7 bits for the network part and 24 bits for the host part, meaning that there can be only 126 class A networks (the values 0 and 127 are reserved), but each of them can accommodate up to 2 - 2 (about 16 million) hosts (again, there are two reserved values). Class B addresses allocate 14 bits for the network and 16 bits for the host, meaning that each class B network has room for 65,534 hosts. Finally, class C addresses have only 8 bits for the host and 21 for the network part. Therefore, a class C network can have only 256 unique host identifiers, which means only 254 attached hosts (one host identifier, 255, is reserved for broadcast, and 0 is not a valid host number). However, the addressing scheme supports 2 class C networks.
On the face of it, this addressing scheme has a lot of flexibility, allowing networks of vastly different sizes to be accommodated fairly efficiently. The original idea was that the Internet would consist of a small number of wide area networks (these would be class A networks), a modest number of site- (campus-) sized networks (these would be class B networks), and a large number of LANs (these would be class C networks). However, it turned out not to be flexible enough, as we will see in a moment. Today, IP addresses are normally "classless"; the details of this are explained below.
Before we look at how IP addresses get used, it is helpful to look at
some practical matters, such as how you write them down. By convention,
IP addresses are written as four decimal integers separated by dots.
Each integer represents the decimal value contained in 1 byte of the
address, starting at the most significant. For example, the address of
the computer on which this sentence was typed is
It is important not to confuse IP addresses with Internet domain names,
which are also hierarchical. Domain names tend to be ASCII strings
separated by dots, such as
cs.princeton.edu. The important thing
about IP addresses is that they are what is carried in the headers of
IP packets, and it is those addresses that are used in IP routers to
make forwarding decisions.
Datagram Forwarding in IP
We are now ready to look at the basic mechanism by which IP routers forward datagrams in an internetwork. Recall from an earlier section that forwarding is the process of taking a packet from an input and sending it out on the appropriate output, while routing is the process of building up the tables that allow the correct output for a packet to be determined. The discussion here focuses on forwarding; we take up routing in a later section.
The main points to bear in mind as we discuss the forwarding of IP datagrams are the following:
Every IP datagram contains the IP address of the destination host.
The network part of an IP address uniquely identifies a single physical network that is part of the larger Internet.
All hosts and routers that share the same network part of their address are connected to the same physical network and can thus communicate with each other by sending frames over that network.
Every physical network that is part of the Internet has at least one router that, by definition, is also connected to at least one other physical network; this router can exchange packets with hosts or routers on either network.
Forwarding IP datagrams can therefore be handled in the following way. A datagram is sent from a source host to a destination host, possibly passing through several routers along the way. Any node, whether it is a host or a router, first tries to establish whether it is connected to the same physical network as the destination. To do this, it compares the network part of the destination address with the network part of the address of each of its network interfaces. (Hosts normally have only one interface, while routers normally have two or more, since they are typically connected to two or more networks.) If a match occurs, then that means that the destination lies on the same physical network as the interface, and the packet can be directly delivered over that network. A later section explains some of the details of this process.
If the node is not connected to the same physical network as the
destination node, then it needs to send the datagram to a router. In
general, each node will have a choice of several routers, and so it
needs to pick the best one, or at least one that has a reasonable chance
of getting the datagram closer to its destination. The router that it
chooses is known as the next hop router. The router finds the correct
next hop by consulting its forwarding table. The forwarding table is
conceptually just a list of
(NetworkNum, NextHop)pairs. (As we will
see below, forwarding tables in practice often
contain some additional information related to the next hop.) Normally,
there is also a default router that is used if none of the entries in
the table matches the destination's network number. For a host, it may
be quite acceptable to have a default router and nothing else—this
means that all datagrams destined for hosts not on the physical network
to which the sending host is attached will be sent out through the
We can describe the datagram forwarding algorithm in the following way:
if (NetworkNum of destination = NetworkNum of one of my interfaces) then deliver packet to destination over that interface else if (NetworkNum of destination is in my forwarding table) then deliver packet to NextHop router else deliver packet to default router
For a host with only one interface and only a default router in its forwarding table, this simplifies to
if (NetworkNum of destination = my NetworkNum) then deliver packet to destination directly else deliver packet to default router
Let's see how this works in the example internetwork of Figure 1. First, suppose that H1 wants to send a datagram to H2. Since they are on the same physical network, H1 and H2 have the same network number in their IP address. Thus, H1 deduces that it can deliver the datagram directly to H2 over the Ethernet. The one issue that needs to be resolved is how H1 finds out the correct Ethernet address for H2—the resolution mechanism described in a later section addresses this issue.
Now suppose H5 wants to send a datagram to H8. Since these hosts are on different physical networks, they have different network numbers, so H5 deduces that it needs to send the datagram to a router. R1 is the only choice—the default router—so H1 sends the datagram over the wireless network to R1. Similarly, R1 knows that it cannot deliver a datagram directly to H8 because neither of R1's interfaces are on the same network as H8. Suppose R1's default router is R2; R1 then sends the datagram to R2 over the Ethernet. Assuming R2 has the forwarding table shown in Table 1, it looks up H8's network number (network 4) and forwards the datagram over the point-to-point network to R3. Finally, R3, since it is on the same network as H8, forwards the datagram directly to H8.
Note that it is possible to include the information about directly connected networks in the forwarding table. For example, we could label the network interfaces of router R2 as interface 0 for the point-to-point link (network 3) and interface 1 for the Ethernet (network 2). Then R2 would have the forwarding table shown in Table 2.
Thus, for any network number that R2 encounters in a packet, it knows what to do. Either that network is directly connected to R2, in which case the packet can be delivered to its destination over that network, or the network is reachable via some next hop router that R2 can reach over a network to which it is connected. In either case, R2 will use ARP, described below, to find the MAC address of the node to which the packet is to be sent next.
The forwarding table used by R2 is simple enough that it could be manually configured. Usually, however, these tables are more complex and would be built up by running a routing protocol such as one of those described in a later section. Also note that, in practice, the network numbers are usually longer (e.g., 128.96).
We can now see how hierarchical addressing—splitting the address into network and host parts—has improved the scalability of a large network. Routers now contain forwarding tables that list only a set of network numbers rather than all the nodes in the network. In our simple example, that meant that R2 could store the information needed to reach all the hosts in the network (of which there were eight) in a four-entry table. Even if there were 100 hosts on each physical network, R2 would still only need those same four entries. This is a good first step (although by no means the last) in achieving scalability.
This illustrates one of the most important principles of building scalable networks: To achieve scalability, you need to reduce the amount of information that is stored in each node and that is exchanged between nodes. The most common way to do that is hierarchical aggregation. IP introduces a two-level hierarchy, with networks at the top level and nodes at the bottom level. We have aggregated information by letting routers deal only with reaching the right network; the information that a router needs to deliver a datagram to any node on a given network is represented by a single aggregated piece of information.
Subnetting and Classless Addressing
The original intent of IP addresses was that the network part would uniquely identify exactly one physical network. It turns out that this approach has a couple of drawbacks. Imagine a large campus that has lots of internal networks and decides to connect to the Internet. For every network, no matter how small, the site needs at least a class C network address. Even worse, for any network with more than 255 hosts, they need a class B address. This may not seem like a big deal, and indeed it wasn't when the Internet was first envisioned, but there are only a finite number of network numbers, and there are far fewer class B addresses than class Cs. Class B addresses tend to be in particularly high demand because you never know if your network might expand beyond 255 nodes, so it is easier to use a class B address from the start than to have to renumber every host when you run out of room on a class C network. The problem we observe here is address assignment inefficiency: A network with two nodes uses an entire class C network address, thereby wasting 253 perfectly useful addresses; a class B network with slightly more than 255 hosts wastes over 64,000 addresses.
Assigning one network number per physical network, therefore, uses up the IP address space potentially much faster than we would like. While we would need to connect over 4 billion hosts to use up all the valid addresses, we only need to connect 2 (about 16,000) class B networks before that part of the address space runs out. Therefore, we would like to find some way to use the network numbers more efficiently.
Assigning many network numbers has another drawback that becomes apparent when you think about routing. Recall that the amount of state that is stored in a node participating in a routing protocol is proportional to the number of other nodes, and that routing in an internet consists of building up forwarding tables that tell a router how to reach different networks. Thus, the more network numbers there are in use, the bigger the forwarding tables get. Big forwarding tables add costs to routers, and they are potentially slower to search than smaller tables for a given technology, so they degrade router performance. This provides another motivation for assigning network numbers carefully.
Subnetting provides a first step to reducing total number of network numbers that are assigned. The idea is to take a single IP network number and allocate the IP addresses with that network number to several physical networks, which are now referred to as subnets. Several things need to be done to make this work. First, the subnets should be close to each other. This is because from a distant point in the Internet, they will all look like a single network, having only one network number between them. This means that a router will only be able to select one route to reach any of the subnets, so they had better all be in the same general direction. A perfect situation in which to use subnetting is a large campus or corporation that has many physical networks. From outside the campus, all you need to know to reach any subnet inside the campus is where the campus connects to the rest of the Internet. This is often at a single point, so one entry in your forwarding table will suffice. Even if there are multiple points at which the campus is connected to the rest of the Internet, knowing how to get to one point in the campus network is still a good start.
The mechanism by which a single network number can be shared among multiple networks involves configuring all the nodes on each subnet with a subnet mask. With simple IP addresses, all hosts on the same network must have the same network number. The subnet mask enables us to introduce a subnet number; all hosts on the same physical network will have the same subnet number, which means that hosts may be on different physical networks but share a single network number. This concept is illustrated in Figure 7.
What subnetting means to a host is that it is now configured with both an IP address and a subnet mask for the subnet to which it is attached. For example, host H1 in Figure 8 is configured with an address of 22.214.171.124 and a subnet mask of 255.255.255.128. (All hosts on a given subnet are configured with the same mask; that is, there is exactly one subnet mask per subnet.) The bitwise AND of these two numbers defines the subnet number of the host and of all other hosts on the same subnet. In this case, 126.96.36.199 AND 255.255.255.128 equals 188.8.131.52, so this is the subnet number for the topmost subnet in the figure.
When the host wants to send a packet to a certain IP address, the first thing it does is to perform a bitwise AND between its own subnet mask and the destination IP address. If the result equals the subnet number of the sending host, then it knows that the destination host is on the same subnet and the packet can be delivered directly over the subnet. If the results are not equal, the packet needs to be sent to a router to be forwarded to another subnet. For example, if H1 is sending to H2, then H1 ANDs its subnet mask (255.255.255.128) with the address for H2 (184.108.40.206) to obtain 220.127.116.11. This does not match the subnet number for H1 (18.104.22.168) so H1 knows that H2 is on a different subnet. Since H1 cannot deliver the packet to H2 directly over the subnet, it sends the packet to its default router R1.
The forwarding table of a router also changes slightly when we introduce
subnetting. Recall that we previously had a forwarding table that
consisted of entries of the form
(NetworkNum, NextHop). To support
subnetting, the table must now hold entries of the form
(SubnetNumber, SubnetMask, NextHop). To find the right entry in the
table, the router ANDs the packet's destination address with the
SubnetMaskfor each entry in turn; if the result matches the
SubnetNumber of the entry, then this is the right entry to use, and
it forwards the packet to the next hop router indicated. In the
example network of Figure 8, router R1 would have the
entries shown in Table 2.
Continuing with the example of a datagram from H1 being sent to H2, R1 would AND H2's address (22.214.171.124) with the subnet mask of the first entry (255.255.255.128) and compare the result (126.96.36.199) with the network number for that entry (188.8.131.52). Since this is not a match, it proceeds to the next entry. This time a match does occur, so R1 delivers the datagram to H2 using interface 1, which is the interface connected to the same network as H2.
We can now describe the datagram forwarding algorithm in the following way:
D = destination IP address for each forwarding table entry (SubnetNumber, SubnetMask, NextHop) D1 = SubnetMask & D if D1 = SubnetNumber if NextHop is an interface deliver datagram directly to destination else deliver datagram to NextHop (a router)
Although not shown in this example, a default route would usually be included in the table and would be used if no explicit matches were found. Note that a naive implementation of this algorithm—one involving repeated ANDing of the destination address with a subnet mask that may not be different every time, and a linear table search—would be very inefficient.
An important consequence of subnetting is that different parts of the internet see the world differently. From outside our hypothetical campus, routers see a single network. In the example above, routers outside the campus see the collection of networks in Figure 8 as just the network 128.96, and they keep one entry in their forwarding tables to tell them how to reach it. Routers within the campus, however, need to be able to route packets to the right subnet. Thus, not all parts of the internet see exactly the same routing information. This is an example of an aggregation of routing information, which is fundamental to scaling of the routing system. The next section shows how aggregation can be taken to another level.
Subnetting has a counterpart, sometimes called supernetting, but more often called Classless Interdomain Routing or CIDR, pronounced "cider." CIDR takes the subnetting idea to its logical conclusion by essentially doing away with address classes altogether. Why isn't subnetting alone sufficient? In essence, subnetting only allows us to split a classful address among multiple subnets, while CIDR allows us to coalesce several classful addresses into a single "supernet." This further tackles the address space inefficiency noted above, and does so in a way that keeps the routing system from being overloaded.
To see how the issues of address space efficiency and scalability of the routing system are coupled, consider the hypothetical case of a company whose network has 256 hosts on it. That is slightly too many for a Class C address, so you would be tempted to assign a class B. However, using up a chunk of address space that could address 65535 to address 256 hosts has an efficiency of only 256/65,535 = 0.39%. Even though subnetting can help us to assign addresses carefully, it does not get around the fact that any organization with more than 255 hosts, or an expectation of eventually having that many, wants a class B address.
The first way you might deal with this issue would be to refuse to give a class B address to any organization that requests one unless they can show a need for something close to 64K addresses, and instead giving them an appropriate number of class C addresses to cover the expected number of hosts. Since we would now be handing out address space in chunks of 256 addresses at a time, we could more accurately match the amount of address space consumed to the size of the organization. For any organization with at least 256 hosts, we can guarantee an address utilization of at least 50%, and typically much more.
Even if you can justify a request of a class B network number, don't bother. They are all spoken for.
This solution, however, raises a problem that is at least as serious: excessive storage requirements at the routers. If a single site has, say, 16 class C network numbers assigned to it, that means every Internet backbone router needs 16 entries in its routing tables to direct packets to that site. This is true even if the path to every one of those networks is the same. If we had assigned a class B address to the site, the same routing information could be stored in one table entry. However, our address assignment efficiency would then be only 6 255 / 65,536 = 6.2%.
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 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
approach used in subnetting, as long as
masks consist of contiguous
bits starting from the most significant bit (which in practice is
almost always the case).
The ability to aggregate routes at the edge of the network as we have just seen is only the first step. Imagine an Internet service provider network, whose primary job is to provide Internet connectivity to a large number of corporations and campuses (customers). If we assign prefixes to the customers in such a way that many different customer networks connected to the provider network share a common, shorter address prefix, then we can get even greater aggregation of routes. Consider the example in Figure 9. Assume that eight customers served by the provider network have each been assigned adjacent 24-bit network prefixes. Those prefixes all start with the same 21 bits. Since all of the customers are reachable through the same provider network, it can advertise a single route to all of them by just advertising the common 21-bit prefix they share. And it can do this even if not all the 24-bit prefixes have been handed out, as long as the provider ultimately will have the right to hand out those prefixes to a customer. One way to accomplish that is to assign a portion of address space to the provider in advance and then to let the network provider assign addresses from that space to its customers as needed. Note that, in contrast to this simple example, there is no need for all customer prefixes to be the same length.
IP Forwarding Revisited
In all our discussion of IP forwarding so far, we have assumed that we could find the network number in a packet and then look up that number in a forwarding table. However, now that we have introduced CIDR, we need to reexamine this assumption. CIDR means that prefixes may be of any length, from 2 to 32 bits. Furthermore, it is sometimes possible to have prefixes in the forwarding table that "overlap," in the sense that some addresses may match more than one prefix. For example, we might find both 171.69 (a 16-bit prefix) and 171.69.10 (a 24-bit prefix) in the forwarding table of a single router. In this case, a packet destined to, say, 184.108.40.206 clearly matches both prefixes. The rule in this case is based on the principle of "longest match"; that is, the packet matches the longest prefix, which would be 171.69.10 in this example. On the other hand, a packet destined to 220.127.116.11 would match 171.69 and not 171.69.10, and in the absence of any other matching entry in the routing table 171.69 would be the longest match.
The task of efficiently finding the longest match between an IP address and the variable-length prefixes in a forwarding table has been a fruitful field of research for many years. The most well-known algorithm uses an approach known as a PATRICIA tree, which was actually developed well in advance of CIDR.
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
18.104.22.168. 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.
A more general solution would be for each host to maintain a table of address pairs; that is, the table would map IP addresses into physical addresses. While this table could be centrally managed by a system administrator and then copied to each host on the network, a better approach would be for each host to dynamically learn the contents of the table using the network. This can be accomplished using the Address Resolution Protocol (ARP). The goal of ARP is to enable each host on a network to build up a table of mappings between IP addresses and link-level addresses. Since these mappings may change over time (e.g., because an Ethernet card in a host breaks and is replaced by a new one with a new address), the entries are timed out periodically and removed. This happens on the order of every 15 minutes. The set of mappings currently stored in a host is known as the ARP cache or ARP table.
ARP takes advantage of the fact that many link-level network technologies, such as Ethernet, support broadcast. If a host wants to send an IP datagram to a host (or router) that it knows to be on the same network (i.e., the sending and receiving nodes have the same IP network number), it first checks for a mapping in the cache. If no mapping is found, it needs to invoke the Address Resolution Protocol over the network. It does this by broadcasting an ARP query onto the network. This query contains the IP address in question (the target IP address). Each host receives the query and checks to see if it matches its IP address. If it does match, the host sends a response message that contains its link-layer address back to the originator of the query. The originator adds the information contained in this response to its ARP table.
The query message also includes the IP address and link-layer address of the sending host. Thus, when a host broadcasts a query message, each host on the network can learn the sender's link-level and IP addresses and place that information in its ARP table. However, not every host adds this information to its ARP table. If the host already has an entry for that host in its table, it "refreshes" this entry; that is, it resets the length of time until it discards the entry. If that host is the target of the query, then it adds the information about the sender to its table, even if it did not already have an entry for that host. This is because there is a good chance that the source host is about to send it an application-level message, and it may eventually have to send a response or ACK back to the source; it will need the source's physical address to do this. If a host is not the target and does not already have an entry for the source in its ARP table, then it does not add an entry for the source. This is because there is no reason to believe that this host will ever need the source's link-level address; there is no need to clutter its ARP table with this information.
Figure 10 shows the ARP packet format for IP-to-Ethernet address mappings. In fact, ARP can be used for lots of other kinds of mappings—the major differences are in the address sizes. In addition to the IP and link-layer addresses of both sender and target, the packet contains
HardwareTypefield, which specifies the type of physical network (e.g., Ethernet)
ProtocolTypefield, which specifies the higher-layer protocol (e.g., IP)
HLen("hardware" address length) and
PLen("protocol" address length) fields, which specify the length of the link-layer address and higher-layer protocol address, respectively
Operationfield, which specifies whether this is a request or a response
The source and target hardware (Ethernet) and protocol (IP) addresses
Note that the results of the ARP process can be added as an extra column in a forwarding table like the one in Table 1. Thus, for example, when R2 needs to forward a packet to network 2, it not only finds that the next hop is R1, but also finds the MAC address to place on the packet to send it to R1.
We have now seen the basic mechanisms that IP provides for dealing with both heterogeneity and scale. On the issue of heterogeneity, IP begins by defining a best-effort service model that makes minimal assumptions about the underlying networks; most notably, this service model is based on unreliable datagrams. IP then makes two important additions to this starting point: (1) a common packet format (fragmentation/reassembly is the mechanism that makes this format work over networks with different MTUs) and (2) a global address space for identifying all hosts (ARP is the mechanism that makes this global address space work over networks with different physical addressing schemes). On the issue of scale, IP uses hierarchical aggregation to reduce the amount of information needed to forward packets. Specifically, IP addresses are partitioned into network and host components, with packets first routed toward the destination network and then delivered to the correct host on that network.
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 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.
Most host operating systems provide a way for a system administrator, or even a user, to manually configure the IP information needed by a host; however, there are some obvious drawbacks to such manual configuration. One is that it is simply a lot of work to configure all the hosts in a large network directly, especially when you consider that such hosts are not reachable over a network until they are configured. Even more importantly, the configuration process is very error prone, since it is necessary to ensure that every host gets the correct network number and that no two hosts receive the same IP address. For these reasons, automated configuration methods are required. The primary method uses a protocol known as the Dynamic Host Configuration Protocol (DHCP).
DHCP relies on the existence of a DHCP server that is responsible for providing configuration information to hosts. There is at least one DHCP server for an administrative domain. At the simplest level, the DHCP server can function just as a centralized repository for host configuration information. Consider, for example, the problem of administering addresses in the internetwork of a large company. DHCP saves the network administrators from having to walk around to every host in the company with a list of addresses and network map in hand and configuring each host manually. Instead, the configuration information for each host could be stored in the DHCP server and automatically retrieved by each host when it is booted or connected to the network. However, the administrator would still pick the address that each host is to receive; he would just store that in the server. In this model, the configuration information for each host is stored in a table that is indexed by some form of unique client identifier, typically the hardware address (e.g., the Ethernet address of its network adaptor).
A more sophisticated use of DHCP saves the network administrator from even having to assign addresses to individual hosts. In this model, the DHCP server maintains a pool of available addresses that it hands out to hosts on demand. This considerably reduces the amount of configuration an administrator must do, since now it is only necessary to allocate a range of IP addresses (all with the same network number) to each network.
Since the goal of DHCP is to minimize the amount of manual configuration required for a host to function, it would rather defeat the purpose if each host had to be configured with the address of a DHCP server. Thus, the first problem faced by DHCP is that of server discovery.
To contact a DHCP server, a newly booted or attached host sends a
DHCPDISCOVER message to a special IP address (255.255.255.255) that is
an IP broadcast address. This means it will be received by all hosts and
routers on that network. (Routers do not forward such packets onto other
networks, preventing broadcast to the entire Internet.) In the simplest
case, one of these nodes is the DHCP server for the network. The server
would then reply to the host that generated the discovery message (all
the other nodes would ignore it). However, it is not really desirable to
require one DHCP server on every network, because this still creates a
potentially large number of servers that need to be correctly and
consistently configured. Thus, DHCP uses the concept of a relay agent.
There is at least one relay agent on each network, and it is configured
with just one piece of information: the IP address of the DHCP server.
When a relay agent receives a
DHCPDISCOVER message, it unicasts it to
the DHCP server and awaits the response, which it will then send back to
the requesting client. The process of relaying a message from a host to
a remote DHCP server is shown in Figure 11.
Figure 12 below shows the format of a DHCP message. The message is actually sent using a protocol called the User Datagram Protocol (UDP) that runs over IP. UDP is discussed in detail in the next chapter, but the only interesting thing it does in this context is to provide a demultiplexing key that says, "This is a DHCP packet."
DHCP is derived from an earlier protocol called BOOTP, and some of the
packet fields are thus not strictly relevant to host configuration. When
trying to obtain configuration information, the client puts its hardware
address (e.g., its Ethernet address) in the
chaddr field. The DHCP
server replies by filling in the
yiaddr ("your" IP address) field and
sending it to the client. Other information such as the default router
to be used by this client can be included in the
In the case where DHCP dynamically assigns IP addresses to hosts, it is clear that hosts cannot keep addresses indefinitely, as this would eventually cause the server to exhaust its address pool. At the same time, a host cannot be depended upon to give back its address, since it might have crashed, been unplugged from the network, or been turned off. Thus, DHCP allows addresses to be leased for some period of time. Once the lease expires, the server is free to return that address to its pool. A host with a leased address clearly needs to renew the lease periodically if in fact it is still connected to the network and functioning correctly.
DHCP illustrates an important aspect of scaling: the scaling of network management. While discussions of scaling often focus on keeping the state in network devices from growing too fast, it is important to pay attention to the growth of network management complexity. By allowing network managers to configure a range of IP addresses per network rather than one IP address per host, DHCP improves the manageability of a network.
Note that DHCP may also introduce some more complexity into network management, since it makes the binding between physical hosts and IP addresses much more dynamic. This may make the network manager's job more difficult if, for example, it becomes necessary to locate a malfunctioning 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, 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.
ICMP also provides the basis for two widely used debugging tools,
ping uses ICMP echo messages to determine if a node
is reachable and alive.
traceroute uses a slightly non-intuitive
technique to determine the set of routers along the path to a
destination, which is the topic for one of the exercises at the end of
Virtual Networks and Tunnels
We conclude our introduction to IP by considering an issue you might not have anticipated, but one that is becoming increasingly important. Our discussion up to this point has focused on making it possible for nodes on different networks to communicate with each other in an unrestricted way. This is the usually the goal in the Internet—everybody wants to be able to send email to everybody, and the creator of a new website wants to reach the widest possible audience. However, there are many situations where more controlled connectivity is required. An important example of such a situation is the virtual private network (VPN).
The term VPN is heavily overused and definitions vary, but intuitively we can define a VPN by considering first the idea of a private network. Corporations with many sites often build private networks by leasing transmission lines from the phone companies and using those lines to interconnect sites. In such a network, communication is restricted to take place only among the sites of that corporation, which is often desirable for security reasons. To make a private network virtual, the leased transmission lines—which are not shared with any other corporations—would be replaced by some sort of shared network. A virtual circuit (VC) is a very reasonable replacement for a leased line because it still provides a logical point-to-point connection between the corporation's sites. For example, if corporation X has a VC from site A to site B, then clearly it can send packets between sites A and B. But there is no way that corporation Y can get its packets delivered to site B without first establishing its own virtual circuit to site B, and the establishment of such a VC can be administratively prevented, thus preventing unwanted connectivity between corporation X and corporation Y.
Figure 13(a) shows two private networks for two separate corporations. In Figure 13(b) they are both migrated to a virtual circuit network. The limited connectivity of a real private network is maintained, but since the private networks now share the same transmission facilities and switches we say that two virtual private networks have been created.
In Figure 13, a virtual circuit network (using ATM, for example) is used to provide the controlled connectivity among sites. It is also possible to provide a similar function using an IP network to provide the connectivity. However, we cannot just connect the various corporations' sites to a single internetwork because that would provide connectivity between corporation X and corporation Y, which we wish to avoid. To solve this problem, we need to introduce a new concept, the IP tunnel.
We can think of an IP tunnel as a virtual point-to-point link between a pair of nodes that are actually separated by an arbitrary number of networks. The virtual link is created within the router at the entrance to the tunnel by providing it with the IP address of the router at the far end of the tunnel. Whenever the router at the entrance of the tunnel wants to send a packet over this virtual link, it encapsulates the packet inside an IP datagram. The destination address in the IP header is the address of the router at the far end of the tunnel, while the source address is that of the encapsulating router.
In the forwarding table of the router at the entrance to the tunnel, this virtual link looks much like a normal link. Consider, for example, the network in Figure 14. A tunnel has been configured from R1 to R2 and assigned a virtual interface number of 0. The forwarding table in R1 might therefore look like Table 4.
|2||Virtual interface 0|
R1 has two physical interfaces. Interface 0 connects to network 1; interface 1 connects to a large internetwork and is thus the default for all traffic that does not match something more specific in the forwarding table. In addition, R1 has a virtual interface, which is the interface to the tunnel. Suppose R1 receives a packet from network 1 that contains an address in network 2. The forwarding table says this packet should be sent out virtual interface 0. In order to send a packet out this interface, the router takes the packet, adds an IP header addressed to R2, and then proceeds to forward the packet as if it had just been received. R2's address is 22.214.171.124; since the network number of this address is 18, not 1 or 2, a packet destined for R2 will be forwarded out the default interface into the internetwork.
Once the packet leaves R1, it looks to the rest of the world like a normal IP packet destined to R2, and it is forwarded accordingly. All the routers in the internetwork forward it using normal means, until it arrives at R2. When R2 receives the packet, it finds that it carries its own address, so it removes the IP header and looks at the payload of the packet. What it finds is an inner IP packet whose destination address is in network 2. R2 now processes this packet like any other IP packet it receives. Since R2 is directly connected to network 2, it forwards the packet on to that network. Figur 14 shows the change in encapsulation of the packet as it moves across the network.
While R2 is acting as the endpoint of the tunnel, there is nothing to prevent it from performing the normal functions of a router. For example, it might receive some packets that are not tunneled, but that are addressed to networks that it knows how to reach, and it would forward them in the normal way.
You might wonder why anyone would want to go to all the trouble of creating a tunnel and changing the encapsulation of a packet as it goes across an internetwork. One reason is security. Supplemented with encryption, a tunnel can become a very private sort of link across a public network. Another reason may be that R1 and R2 have some capabilities that are not widely available in the intervening networks, such as multicast routing. By connecting these routers with a tunnel, we can build a virtual network in which all the routers with this capability appear to be directly connected. A third reason to build tunnels is to carry packets from protocols other than IP across an IP network. As long as the routers at either end of the tunnel know how to handle these other protocols, the IP tunnel looks to them like a point-to-point link over which they can send non-IP packets. Tunnels also provide a mechanism by which we can force a packet to be delivered to a particular place even if its original header—the one that gets encapsulated inside the tunnel header—might suggest that it should go somewhere else. Thus, we see that tunneling is a powerful and quite general technique for building virtual links across internetworks. So general, in fact, that the technique recurses, with the most common use case being to tunnel IP over IP.
Tunneling does have its downsides. One is that it increases the length of packets; this might represent a significant waste of bandwidth for short packets. Longer packets might be subject to fragmentation, which has its own set of drawbacks. There may also be performance implications for the routers at either end of the tunnel, since they need to do more work than normal forwarding as they add and remove the tunnel header. Finally, there is a management cost for the administrative entity that is responsible for setting up the tunnels and making sure they are correctly handled by the routing protocols.