Implementation of New Computer Network
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Here we are going to implement an new computer network for this company that 25 employees have been working in.
Suppose you want to build a computer network, one that has potential to grow to global proportions to support applications as diverse as teleconferencing, video-on-demand, electronic commerce, distributed computing, and digital libraries. What available technologies would serve as the underlying building blocks, and what kind of software architecture would you design t integrate these building blocks into an effective communication service?
Suppose you want to build a computer network, one that has the potential togrow to global proportions and to support applications as diverse as teleconferencing, video-on-demand, electronic commerce, distributed computing, and digital libraries. What available technologies would serve as the underlying building blocks, and what kind of software architecture would you design to integrate these building blocks into an effective communication service? Answering this question is the overriding goal of — to describe the available building materials and then to show how they can be used to construct a network from the ground up.
Before we can understand how to design a computer network, we should first agree on exactly what a computer network is. At one time, the term network meant the set of serial lines used to attach dumb terminals to mainframe computers. To some, the term implies the voice telephone network. To others, the only interesting network is the cable network used to disseminate video signals. The main thing these networks have in common is that they are specialized to handle one particular kind of data (keystrokes, voice, or video) and they typically connect to special-purpose devices (terminals, hand receivers, and television sets).
What distinguishes a computer network from these other types of networks? Probably the most important characteristic of a computer network is its generality. Computer networks are built primarily from general-purpose programmable hardware, and they are not optimized for a particular application like making phone calls or delivering television signals. Instead, they are able to carry many different types of data, and they support a wide, and ever-growing, range of applications. This chapter looks at some typical applications of computer networks and discusses the requirements that a network designer who wishes to support such applications must be aware of.
Once we understand the requirements, how do we proceed? Fortunately, we will not be building the first network. Others, most notably the community of researchers responsible for the Internet, have gone before us. We will use the wealth of experience generated from the Internet to guide our design. This experience is embodied in a network architecture that identifies the available hardware and software components and shows how they can be arranged to form a complete network system.
To start us on the road toward understanding how to build a network, this chapter does four things. First, it explores the requirements that different applications and different communities of people (such as network users and network operators) place on the network. Second, it introduces the idea of a network architecture, which lays the foundation for the rest of the book. Third, it introduces some of the key elements in the implementation of computer networks. Finally, it identifies the key metrics that are used to evaluate the performance of computer networks.
Most people know the Internet through its applications: the World Wide Web, email, streaming audio and video, chat rooms, and music (file) sharing. The Web, for example, presents an intuitively simple interface. Users view pages full of textual and graphical objects, click on objects that they want to learn more about, and a corresponding new page appears. Most people are also aware that just under the covers, each selectable object on a page is bound to an identifier for the next page to be viewed. This identifier, called a Uniform Resource Locator (URL), is used to provide a way of identifying all the possible pages that can be viewed from your web browser. For example,
is the URL for a page providing information about one of this book's authors: the string http indicates that the HyperText Transfer Protocol (HTTP) should be used to download the page, www.cs.princeton.edu is the name of the machine that serves the page, and
uniquely identifies Larry's home page at this site. What most Web users are not aware of, however, is that by clicking on just one such URL, as many as 17 messages may be exchanged over the Internet, and this assumes the page itself is small enough to fit in a single message. This number includes up to six messages to translate the server name (www.cs.princeton.edu) into its Internet address (22.214.171.124), three messages to set up a Transmission Control Protocol (TCP) connection between your browser and this server, four messages for your browser to send the HTTP "get" request and the server to respond with the requested page (and for each side to acknowledge receipt of that message), and four messages to tear down the TCP connection. Of course, this does not include the millions of messages exchanged by Internet nodes throughout the day, just to let each other know that they exist and are ready to serve web pages, translate names to addresses, and forward messages toward their ultimate destination.
Another widespread application of the Internet is the delivery of "streaming" audio and video. While an entire video file could first be fetched from a remote machine and then played on the local machine, similar to the process of downloading and displaying a web page, this would entail waiting for the last second of the video file to be delivered before starting to look at it. Streaming video implies that the sender and the receiver are, respectively, the source and the sink for the video stream. That is, the source generates a video stream (perhaps using a video capture card), sends it across the Internet in messages, and the sink displays the stream as it arrives.
There are a variety of different classes of video applications. One class of video application is video-on-demand, which reads a pre-existing movie from disk and transmits it over the network. Another kind of application is videoconferencing, which is in some ways the more challenging (and, for networking people, interesting) case because it has very tight timing constraints. Just as when using the telephone, the interactions among the participants must be timely. When a person at one end gestures, then that action must be displayed at the other end as quickly as possible. Too much delay makes the system unusable. Contrast this with video-on-demand where, if it takes several seconds from the time the user starts the video until the first image is displayed, the service is still deemed satisfactory. Also, interactive video usually implies that video is flowing in both directions, while a video-on-demand application is most likely sending video in only one direction.
One pioneering example of a videoconferencing tool, developed in the early and mid-1990s, is vic. shows the control panel for a vic session. vic is actually one of a suite of conferencing tools designed at Lawrence Berkeley Laboratory and UC Berkeley. The others include a whiteboard application (wb) that allows users to send sketches and slides to each other, a visual audio tool called vat, and a session directory (sdr) that is used to create and advertise videoconferences. All these tools run on Unix—hence their lowercase names—and are freely available on the Internet. Many similar tools are available for other operating systems. It is interesting to note that while video over the Internet is still considered to be in its relative infancy at the time of this writing (2006), that the tools to support video over IP have existed for well over a decade.
Although they are just two examples, downloading pages from the Web and participating in a videoconference demonstrate the diversity of applications that can be built on top of the Internet, and hint at the complexity of the Internet's design. Starting from the beginning, and addressing one problem at time, the rest of this book explains how to build a network that supports such a wide range of applications. Chapter 9 concludes the book by revisiting these two specific applications, as well as several others that have become popular on today's Internet.
We have just established an ambitious goal for ourselves: to understand how to build a computer network from the ground up. Our approach to accomplishing this goal will be to start from first principles, and then ask the kinds of questions we would naturally ask if building an actual network. At each step, we will use today's protocols to illustrate various design choices available to us, but we will not accept these existing artifacts as gospel. Instead, we will be asking (and answering) the question of why networks are designed the way they are. While it is tempting to settle for just understanding the way it's done today, it is important to recognize the underlying concepts because networks are constantly changing as the technology evolves and new applications are invented. It is our experience that once you understand the fundamental ideas, any new protocol that you are confronted with will be relatively easy to digest.
The first step is to identify the set of constraints and requirements that influence network design. Before getting started, however, it is important to understand that the expectations you have of a network depend on your perspective:
- An application programmer would list the services that his application needs, for example, a guarantee that each message the application sends will be delivered without error within a certain amount of time.
- A network designer would list the properties of a cost-effective design, for example, that network resources are efficiently utilized and fairly allocated to different users.
- A network provider would list the characteristics of a system that is easy to administer and manage, for example, in which faults can be easily isolated and whereitiseasytoaccountfor usage.
This section attempts to distill these different perspectives into a high-level introduction to the major considerations that drive network design, and in doing so, identifies the challenges addressed throughout the rest of this book.
Starting with the obvious, a network must provide connectivity among a set of computers. Sometimes it is enough to build a limited network that connects only a few select machines. In fact, for reasons of privacy and security, many private (corporate) networks have the explicit goal of limiting the set of machines that are connected. In contrast, other networks (of which the Internet is the prime example) are designed to grow in a way that allows them the potential to connect all the computers in the world. A system that is designed to support growth to an arbitrarily large size is said to scale. Using the Internet as a model, this book addresses the challenge of scalability.
Links, Nodes, and Clouds
Network connectivity occurs at many different levels. At the lowest level, a network can consist of two or more computers directly connected by some physical medium, such as a coaxial cable or an optical fiber. We call such a physical medium a link,and we often refer to the computers it connects as nodes. (Sometimes a node is a more specialized piece of hardware rather than a computer, but we overlook that distinction for the purposes of this discussion.) As illustrated in, physical links are sometimes limited to a pair of nodes (such a link is said to be point-to-point), while in other cases, more than two nodes may share a single physical link (such a link is said to be multiple-access). Whether a given link supports point-to-point or multiple-access connectivity depends on how the node is attached to the link. It is also the case that multiple-access links are often limited in size, in terms of both the geographical distance they can cover and the number of nodes they can connect.
If computer networks were limited to situations in which all nodes are directly connected to each other over a common physical medium, then networks would either be very limited in the number of computers they could connect, or the number of wires coming out of the back of each node would quickly become both unmanageable and very expensive. Fortunately, connectivity between two nodes does not necessarily imply a direct physical connection between them—indirect connectivity may be achieved among a set of cooperating nodes. Consider the following two examples of how a collection of computers can be indirectly connected.
shows a set of nodes, each of which is attached to one or more point- to-point links. Those nodes that are attached to at least two links run software that forwards data received on one link out on another. If organized in a systematic way, these forwarding nodes form a switched network. There are numerous types of switched networks, of which the two most common are circuit-switched and packet-switched. The former is most notably employed by the telephone system, while the latter is used for the overwhelming majority of computer networks and will be the focus of this book. The important feature of packet-switched networks is that the nodes in such a network send discrete blocks of data to each other. Think of these blocks of data as corresponding to some piece of application data such as a file, a piece of email, or an image. We call each block of data either a packet or a message, and for now we use these terms interchangeably; we discuss the reason they are not always the same in Section 1.2.2.
Packet-switched networks typically use a strategy called store-and-forward. As the name suggests, each node in a store-and-forward network first receives a complete packet over some link, stores the packet in its internal memory, and then forwards the complete packet to the next node. In contrast, a circuit-switched network first establishes a dedicated circuit across a sequence of links and then allows the source node to send a stream of bits across this circuit to a destination node. The major reason for using packet switching rather than circuit switching in a computer network is efficiency, discussed in the next subsection.
The cloud in distinguishes between the nodes on the inside that implement the network (they are commonly called switches, and their primary function is to store and forward packets) and the nodes on the outside of the cloud that use the network (they are commonly called hosts, and they support users and run application programs). Also note that the cloud in is one of the most important icons of computer networking. In general, we use a cloud to denote any type of network, whether it is a single point-to-point link, a multiple-access link, or a switched network. Thus, whenever you see a cloud used in a figure, you can think of it as a placeholder for any of the networking technologies covered in this book.
A second way in which a set of computers can be indirectly connected is shown in . In this situation, a set of independent networks (clouds) are interconnected to form an internetwork, or internet for short. We adopt the Internet's convention of referring to a generic internetwork of networks as a lowercase i internet, and the currently operational TCP/IP Internet as the capital I Internet. A node that is connected to two or more networks is commonly called a router or gateway, and it plays much the same role as a switch—it forwards messages from one network to another. Note that an internet can itself be viewed as another kind of network, which means that an internet can be built from an interconnection of internets. Thus, we can recursively build arbitrarily large networks by interconnecting clouds to form larger clouds.
Just because a set of hosts are directly or indirectly connected to each other does not mean that we have succeeded in providing host-to-host connectivity. The final requirement is that each node must be able to state which of the other nodes on the network it wants to communicate with. This is done by assigning an address to each node. An address is a byte string that identifies a node; that is, the network can use a node's address to distinguish it from the other nodes connected to the network. When a source node wants the network to deliver a message to a certain destination node, it specifies the address of the destination node. If the sending and receiving nodes are not directly connected, then the switches and routers of the network use this address to decide how to forward the message toward the destination. The process of determining systematically how to forward messages toward the destination node based on its address is called routing.
This brief introduction to addressing and routing has presumed that the source node wants to send a message to a single destination node (unicast). While this is the most common scenario, it is also possible that the source node might want to broadcast a message to all the nodes on the network. Or a source node might want to send a message to some subset of the other nodes, but not all of them, a situation called multicast. Thus, in addition to node-specific addresses, another requirement of a network is that it supports multicast and broadcast addresses.
The main idea to take away from this discussion is that we can define a network recursively as consisting of two or more nodes connected by a physical link, or as two or more networks connected by a node. In other words, a network can be constructed from a nesting of networks, where at the bottom level, the network is implemented by some physical medium. One of the key challenges in providing network connectivity is to define an address for each node that is reachable on the network (including support for broadcast and multicast connectivity), and to be able to use this address to route messages toward the appropriate destination node(s).
1.2.2 Cost-Effective Resource Sharing
As stated above, this book focuses on packet-switched networks. This section explains the key requirement of computer networks—efficiency—that leads us to packet switching as the strategy of choice.
Given a collection of nodes indirectly connected by a nesting of networks, it is possible for any pair of hosts to send messages to each other across a sequence of links and nodes. Of course, we want to do more than support just one pair of communicating hosts—we want to provide all pairs of hosts with the ability to exchange messages. The question, then, is how do all the hosts that want to communicate share the network, especially if they want to use it at the same time? And, as if that problem isn't hard enough, how do several hosts share the same link when they all want to use it at the same time?
To understand how hosts share a network, we need to introduce a fundamental concept, multiplexing, which means that a system resource is shared among multiple users. At an intuitive level, multiplexing can be explained by analogy to a timesharing computer system, where a single physical CPU is shared (multiplexed) among multiple jobs, each of which believes it has its own private processor. Similarly, data being sent by multiple users can be multiplexed over the physical links that make up a network.
To see how this might work, consider the simple network illustrated in , where the three hosts on the left side of the network (senders S1'S3) are sending data to the three hosts on the right (receivers R1'R3) by sharing a switched network that contains only one physical link. (For simplicity, assume that host S1 is sending data to host R1, and so on.) In this situation, three flows of data—corresponding to the three pairs of hosts—are multiplexed onto a single physical link by switch 1 and then demultiplexed back into separate flows by switch 2. Note that we are being intentionally vague about exactly what a "flow of data" corresponds to. For the purposes of this discussion, assume that each host on the left has a large supply of data that it wants to send to its counterpart on the right.
There are several different methods for multiplexing multiple flows onto one physical link. One common method is synchronous time-division multiplexing (STDM). The idea of STDM is to divide time into equal-sized quanta and, in a round-robin fashion, give each flow a chance to send its data over the physical link. In other words, during time quantum 1, data from S1 to R1 is transmitted; during time quantum 2, data from S2 to R2 is transmitted; in quantum 3, S3 sends data to R3. At this point, the first flow (S1 to R1) gets to go again, and the process repeats. Another method is frequency-division multiplexing (FDM). The idea of FDM is to transmit each flow over the physical link at a different frequency, much the same way that the signals for different TV stations are transmitted at a different frequency on a physical cable TV link.
Although simple to understand, both STDM and FDM are limited in two ways. First, if one of the flows (host pairs) does not have any data to send, its share of the physical link—that is, its time quantum or its frequency—remains idle, even if one of the other flows has data to transmit. For example, S3 had to wait its turn behind S1 and S2 in the previous paragraph, even if S1 and S2 had nothing to send. For computer communication, the amount of time that a link is idle can be very large—for example, consider the amount of time you spend reading a web page (leaving the link idle) compared to the time you spend fetching the page. Second, both STDM and FDM are limited to situations in which the maximum number of flows is fixed and known ahead of time. It is not practical to resize the quantum or to add additional quanta in the case of STDM or to add new frequencies in the case of FDM.
The form of multiplexing that we make most use of in this book is called statistical multiplexing. Although the name is not all that helpful for understanding the concept, statistical multiplexing is really quite simple, with two key ideas. First, it is like STDM in that the physical link is shared over time—first data from one flow is transmitted over the physical link, then data from another flow is transmitted, and so on. Unlike STDM, however, data is transmitted from each flow on demand rather than during a predetermined time slot. Thus, if only one flow has data to send, it gets to transmit that data without waiting for its quantum to come around and thus without having to watch the quanta assigned to the other flows go by unused. It is this avoidance of idle time that gives packet switching its efficiency.
As defined so far, however, statistical multiplexing has no mechanism to ensure that all the flows eventually get their turn to transmit over the physical link. That is, once a flow begins sending data, we need some way to limit the transmission, so that the other flows can have a turn. To account for this need, statistical multiplexing defines an upper bound on the size of the block of data that each flow is permitted to transmit at a given time. This limited-size block of data is typically referred to as a packet, to distinguish it from the arbitrarily large message that an application program might want to transmit. Because a packet-switched network limits the maximum size of packets, a host may not be able to send a complete message in one packet. The source may need to fragment the message into several packets, with the receiver reassembling the packets back into the original message.
In other words, each flow sends a sequence of packets over the physical link, with a decision made on a packet-by-packet basis as to which flow's packet to send next. Notice that if only one flow has data to send, then it can send a sequence of packets back-to-back. However, should more than one of the flows have data to send, then their packets are interleaved on the link. depicts a switch multiplexing packets from multiple sources onto a single shared link.
The decision as to which packet to send next on a shared link can be made in a number of different ways. For example, in a network consisting of switches interconnected by links such as the one in the decision would be made by the switch that transmits packets onto the shared link. (As we will see later, not all packet-switched networks actually involve switches, and they may use other mechanisms to determine whose packet goes onto the link next.) Each switch in a packet-switched network makes this decision independently, on a packet-by-packet basis. One of the issues that faces a network designer is how to make this decision in a fair manner. For example, a switch could be designed to service packets on a first-in-first-out (FIFO) basis. Another approach would be to transmit the packets from each of the different flows that are currently sending data through the switch in a round-robin manner. This might be done to ensure that certain flows receive a particular share of the link's bandwidth, or that they never have their packets delayed in the switch for more than a certain length of time. A network that attempts to allocate bandwidth to particular flows is sometimes said to support quality of service (QoS), a topic that we return to in Chapter 6.
Also, notice in that since the switch has to multiplex three incoming packet streams onto one outgoing link, it is possible that the switch will receive packets faster than the shared link can accommodate. In this case, the switch is forced to buffer these packets in its memory. Should a switch receive packets faster than it can send them for an extended period of time, then the switch will eventually run out of buffer space, and some packets will have to be dropped. When a switch is operating in this state, it is said to be congested.
The bottom line is that statistical multiplexing defines a cost-effective way for multiple users (e.g., host-to-host flows of data) to share network resources (links and nodes) in a fine-grained manner. It defines the packet as the granularity with which the links of the network are allocated to different flows, with each switch able to schedule the use of the physical links it is connected to on a per-packet basis. Fairly allocating link capacity to different flows and dealing with congestion when it occurs are the key challenges of statistical multiplexing.
1.2.3 Support for Common Services
While the previous section outlined the challenges involved in providing costeffective connectivity among a group of hosts, it is overly simplistic to view a computer network as simply delivering packets among a collection of computers. It is more accurate to think of a network as providing the means for a set of application processes that are distributed over those computers to communicate. In other words, the next requirement of a computer network is that the application programs running on the hosts connected to the network must be able to communicate in a meaningful way.
SANs, LANs, MANs, and WANs
When two application programs need to communicate with each other, there are a lot of complicated things that need to happen beyond simply sending a message from one host to another. One option would be for application designers to build all that complicated functionality into each application program. However, since many applications need common services, it is much more logical to implement those common services once and then to let the application designer build the application using those services. The challenge for a network designer is to identify the right set of common services. The goal is to hide the complexity of the network from the application without overly constraining the application designer.
Intuitively, we view the network as providing logical channels over which application-level processes can communicate with each other; each channel provides the set of services required by that application. In other words, just as we use a cloud to abstractly represent connectivity among a set of computers, we now think of a channel as connecting one process to another. shows a pair of application-level processes communicating over a logical channel that is, in turn, implemented on top of a cloud that connects a set of hosts. We can think of the channel as being like a pipe connecting two applications, so that a sending application can put data in one end and expect that data to be delivered by the network to the application at the other end of the pipe.
Thechallengeistorecognize what functionality the channels should provide to application programs. For example, does the application require a guarantee that messages sent over the channel are delivered, or is it acceptable if some messages fail to arrive? Is it necessary that messages arrive at the recipient process in the same order in which they are sent, or does the recipient not care about the order in which messages arrive? Does the network need to ensure that no third parties are able to eavesdrop on the channel, or is privacy not a concern? In general, a network provides a variety of different types of channels, with each application selecting the type that best meets its needs. The rest of this section illustrates the thinking involved in defining useful channels.
Identifying Common Communication Patterns
Designing abstract channels involves first understanding the communication needs of a representative collection of applications, then extracting their common communication requirements, and finally incorporating the functionality that meets these requirements in the network.
One of the earliest applications supported on any network is a file access program like FTP (File Transfer Protocol) or NFS (Network File System). Although many details vary—for example, whether whole files are transferred across the network or only single blocks of the file are read/written at a given time—the communication component of remote file access is characterized by a pair of processes, one that requests that a file be read or written and a second process that honors this request. The process that requests access to the file is called the client, and the process that supports access to the file is called the server.
Reading a file involves the client sending a small request message to a server and the server responding with a large message that contains the data in the file. Writing works in the opposite way—the client sends a large message containing the data to be written to the server, and the server responds with a small message confirming that the write to disk has taken place. A digital library, as exemplified by the World Wide Web, is another application that behaves in a similar way: a client process makes a request, and a server process responds by returning the requested data.
Using file access, a digital library, and the two video applications described in the Preface (videoconferencing and video-on-demand) as a representative sample, we might decide to provide the following two types of channels: request/reply channels and message stream channels. The request/reply channel would be used by the file transfer and digital library applications. It would guarantee that every message sent by one side is received by the other side and that only one copy of each message is delivered. The request/reply channel might also protect the privacy and integrity of the data that flows over it, so that unauthorized parties cannot read or modify the data being exchanged between the client and server processes.
The message stream channel could be used by both the video-on-demand and videoconferencing applications, provided it is parameterized to support both one-way and two-way traffic and to support different delay properties. The message stream channel might not need to guarantee that all messages are delivered, since a video application can operate adequately even if some video frames are not received. It would, however, need to ensure that those messages that are delivered arrive in the same order in which they were sent, to avoid displaying frames out of sequence. Like the request/reply channel, the message stream channel might want to ensure the privacy and integrity of the video data. Finally, the message stream channel might need to support multicast, so that multiple parties can participate in the teleconference or view the video.
While it is common for a network designer to strive for the smallest number of abstract channel types that can serve the largest number of applications, there is a danger in trying to get away with too few channel abstractions. Simply stated, if you have a hammer, then everything looks like a nail. For example, if all you have are message stream and request/reply channels, then it is tempting to use them for the next application that comes along, even if neither type provides exactly the semantics needed by the application. Thus, network designers will probably be inventing new types of channels— and adding options to existing channels—for as long as application programmers are inventing new applications.
Also note that independent of exactly what functionality a given channel provides, there is the question of where that functionality is implemented. In many cases, it is easiest to view the host-to-host connectivity of the underlying network as simply providing a bit pipe, with any high-level communication semantics provided at the end hosts. The advantage of this approach is it keeps the switches in the middle of the network as simple as possible—they simply forward packets—but it requires the end hosts to take on much of the burden of supporting semantically rich process-to-process channels. The alternative is to push additional functionality onto the switches, thereby allowing the end hosts to be "dumb" devices (e.g., telephone handsets). We will see this question of how various network services are partitioned between the packet switches and the end hosts (devices) as a recurring issue in network design.
As suggested by the examples just considered, reliable message delivery is one of the most important functions that a network can provide. It is difficult to determine how to provide this reliability, however, without first understanding how networks can fail. The first thing to recognize is that computer networks do not exist in a perfect world. Machines crash and later are rebooted, fibers are cut, electrical interference corrupts bits in the data being transmitted, switches run out of buffer space, and if these sorts of physical problems aren't enough to worry about, the software that manages the hardware sometimes forwards packets into oblivion. Thus, a major requirement of a network is to recover from certain kinds of failures, so that application programs don't have to deal with them, or even be aware of them.
There are three general classes of failure that network designers have to worry about. First, as a packet is transmitted over a physical link, bit errors may be introduced into the data; that is, a 1 is turned into a 0 or vice versa. Sometimes single bits are corrupted, but more often than not, a burst error occurs—several consecutive bits are corrupted. Bit errors typically occur because outside forces, such as lightning strikes, power surges, and microwave ovens, interfere with the transmission of data. The good news is that such bit errors are fairly rare, affecting on average only one out of every 106 to 107 bits on a typical copper-based cable and one out of every 1012 to 1014 bits on a typical optical fiber. As we will see, there are techniques that detect these bit errors with high probability. Once detected, it is sometimes possible to correct for such errors—if we know which bit or bits are corrupted, we can simply flip them—while in other cases the damage is so bad that it is necessary to discard the entire packet. In such a case, the sender may be expected to retransmit the packet.
The second class of failure is at the packet, rather than the bit, level; that is, a complete packet is lost by the network. One reason this can happen is that the packet contains an uncorrectable bit error and therefore has to be discarded. A more likely reason, however, is that one of the nodes that has to handle the packet—for example, a switch that is forwarding it from one link to another—is so overloaded that it has no place to store the packet, and therefore is forced to drop it. This is the problem of congestion mentioned in Section 1.2.2. Less commonly, the software running on one of the nodes that handles the packet makes a mistake. For example, it might incorrectly forward a packet out on the wrong link, so that the packet never finds its way to the ultimate destination. As we will see, one of the main difficulties in dealing with lost packets is distinguishing between a packet that is indeed lost and one that is merely late in arriving at the destination.
The third class of failure is at the node and link level; that is, a physical link is cut, or the computer it is connected to crashes. This can be caused by software that crashes, a power failure, or a reckless backhoe operator. Failures due to misconfiguration of a network device are also common. While any of these failures can eventually be corrected, they can have a dramatic effect on the network for an extended period of time. However, they need not totally disable the network. In a packet-switched network, for example, it is sometimes possible to route around a failed node or link. One of the difficulties in dealing with this third class of failure is distinguishing between a failed computer and one that is merely slow, or in the case of a link, between one that has been cut and one that is very flaky and therefore introducing a high number of bit errors.
The key idea to take away from this discussion is that defining useful channels involves both understanding the applications' requirements and recognizing the limitations of the underlying technology. The challenge is to fill in the gap between what the application expects and what the underlying technology can provide. This is sometimes called the semantic gap.
1.3 NETWORK ARCHITECTURE
In case you hadn't noticed, the previous section established a pretty substantial set of requirements for network design—a computer network must provide general, costeffective, fair, and robust connectivity among a large number of computers. As if this weren't enough, networks do not remain fixed at any single point in time, but must evolve to accommodate changes in both the underlying technologies upon which they are based as well as changes in the demands placed on them by application programs. Designing a network to meet these requirements is no small task.
To help deal with this complexity, network designers have developed general blueprints—usually called network architectures—that guide the design and implementation of networks. This section defines more carefully what we mean by a network architecture by introducing the central ideas that are common to all network architectures. It also introduces two of the most widely referenced architectures—the OSI architecture and the Internet architecture.
1.3.1 Layering and Protocols
When a system gets complex, the system designer introduces another level of abstraction. The idea of an abstraction is to define a unifying model that can capture some important aspect of the system, encapsulate this model in an object that provides an interface that can be manipulated by other components of the system, and hide the details of how the object is implemented from the users of the object. The challenge is to identify abstractions that simultaneously provide a service that proves useful in a large number of situations and that can be efficiently implemented in the underlying system. This is exactly what we were doing when we introduced the idea of a channel in the previous section: We were providing an abstraction for applications that hides the complexity of the network from application writers.
Abstractions naturally lead to layering, especially in network systems. The general idea is that you start with the services offered by the underlying hardware, and then add a sequence of layers, each providing a higher (more abstract) level of service. The services provided at the high layers are implemented in terms of the services provided by the low layers. Drawing on the discussion of requirements given in the previous section, for example, we might imagine a simple network as having two layers of abstraction sandwiched between the application program and the underlying hardware, as illustrated in . The layer immediately above the hardware in this case might provide host- to-host connectivity, abstracting away the fact that there may be an arbitrarily complex network topology between any two hosts. The next layer up builds on the available host- to-host communication service and provides support for process-to-process channels, abstracting away the fact that the network occasionally loses messages, for example.
Layering provides two nice features. First, it decomposes the problem of building a network into more manageable components. Rather than implementing a monolithic piece of software that does everything you will ever want, you can implement several layers, each of which solves one part of the problem. Second, it provides a more modular design. If you decide that you want to add some new service, you may only need to modify the functionality at one layer, reusing the functions provided at all the other layers.
Thinking of a system as a linear sequence of layers is an oversimplification, however. Many times there are multiple abstractions provided at any given level of the system, each providing a different service to the higher layers but building on the same low-level abstractions. To see this, consider the two types of channels discussed in Section 1.2.3: One provides a request/reply service and one supports a message stream service. These two channels might be alternative offerings at some level of a multilevel networking system, as illustrated in .
Using this discussion of layering as a foundation, we are now ready to discuss the architecture of a network more precisely. For starters, the abstract objects that make up the layers of a network system are called protocols. That is, a protocol provides a communication service that higher-level objects (such as application processes, or perhaps higher-level protocols) use to exchange messages. For example, we could imagine a network that supports a request/reply protocol and a message stream protocol, corresponding to the request/reply and message stream channels discussed above.
Each protocol defines two different interfaces. First, it defines a service interface to the other objects on the same computer that want to use its communication services. This service interface defines the operations that local objects can perform on the protocol. For example, a request/reply protocol would support operations by which an application can send and receive messages. An implementation of the HTTP protocol could support an operation to fetch a page of hypertext from a remote server. An application such as a web browser would invoke such an operation whenever the browser needs to obtain a new page, for example, when the user clicks on a link in the currently displayed page.
Second, a protocol defines a peer interface to its counterpart (peer) on another machine. This second interface defines the form and meaning of messages exchanged between protocol peers to implement the communication service. This would determine the way in which a request/reply protocol on one machine communicates with its peer on another machine. In the case of HTTP, for example, the protocol specification defines in detail how a "GET" command is formatted, what arguments can be used with the command, and how a web server should respond when it receives such a command. (We will look more closely at this particular protocol in Section 9.1.2.)
To summarize, a protocol defines a communication service that it exports locally (the service interface), along with a set of rules governing the messages that the protocol exchanges with its peer(s) to implement this service (the peer interface). This situation is illustrated in .
Except at the hardware level where peers directly communicate with each other over a link, peer-to-peer communication is indirect—each protocol communicates with its peer by passing messages to some lower-level protocol, which in turn delivers the message to its peer. In addition, there are potentially multiple protocols at any given level, each providing a different communication service. We therefore represent the suite of protocols that make up a network system with a protocol graph. The nodes of the graph correspond to protocols, and the edges represent a dependson relation. For example, illustrates a protocol graph for the hypothetical layered system we have been discussing—the protocols Request/Reply Protocol (RRP) and Message Stream Protocol (MSP) implement two different types of process-to-process channels, and both depend on Host-to-Host Protocol (HHP), which provides a host-to-host connectivity service.
In this example, suppose that the file access program on host 1 wants to send a message to its peer on host 2 using the communication service offered by protocol RRP. In this case, the file application asks RRP to send the message on its behalf. To communicate with its peer, RRP then invokes the services of HHP, which in turn transmits the message to its peer on the other machine. Once the message has arrived at protocol HHP on host 2, HHP passes the message up to RRP, which in turn delivers the message to the file application. In this particular case, the application is said to employ the services of the protocol stack RRP/HHP.
Note that the term protocol is used in two different ways. Sometimes it refers to the abstract interfaces—that is, the operations defined by the service interface and the form and meaning of messages exchanged between peers—and sometimes it refers to the module that actually implements these two interfaces. To distinguish between the interfaces and the module that implements these interfaces, we generally refer to the former as a protocol specification. Specifications are generally expressed using a combination of prose, pseudocode, state transition diagrams, pictures of packet formats, and other abstract notations. It should be the case that a given protocol can be implemented in different ways by different programmers, as long as each adheres to the specification. The challenge is ensuring that two different implementations of the same specification can successfully exchange messages. Two or more protocol modules that do accurately implement a protocol specification are said to interoperate with each other.
We can imagine many different protocols and protocol graphs that satisfy the communication requirements of a collection of applications. Fortunately, there exist standardization bodies, such as the International Standards Organization (ISO) and the Internet Engineering Task Force (IETF), that establish policies for a particular protocol graph. We call the set of rules governing the form and content of a protocol graph a network architecture. Although beyond the scope of this book, standardization bodies such as the ISO and the IETF have established well-defined procedures for introducing, validating, and finally approving protocols in their respective architectures. We briefly describe the architectures defined by the ISO and the IETF shortly, but first there are two additional things we need to explain about the mechanics of a protocol graph.
Consider what happens in when one of the application programs sends a message to its peer by passing the message to protocol RRP. From RRP's perspective, the message it is given by the application is an uninterpreted string of bytes. RRP does not care that these bytes represent an array of integers, an email message, a digital image, or whatever; it is simply charged with sending them to its peer. However, RRP must communicate control information to its peer, instructing it how to handle the message when it is received. RRP does this by attaching a header to the message. Generally speaking, a header is a small data structure—from a few bytes to a few dozen bytes—that is used among peers to communicate with each other. As the name suggests, headers are usually attached to the front of a message. In some cases, however, this peer-to-peer control information is sent at the end of the message, in which case it is called a trailer. The exact format for the header attached by RRP is defined by its protocol specification. The rest of the message—that is, the data being transmitted on behalf of the application—is called the message's body or payload. We say that the application's data is encapsulated in the new message created by protocol RRP.
This process of encapsulation is then repeated at each level of the protocol graph; for example, HHP encapsulates RRP's message by attaching a header of its own. If we now assume that HHP sends the message to its peer over some network, then when the message arrives at the destination host, it is processed in the opposite order: HHP first interprets the HHP header at the front of the message (i.e., takes whatever action is appropriate given the contents of the header), and passes the body of the message (but not the HHP header) up to RRP, which takes whatever action is indicated by the RRP header that its peer attached, and passes the body of the message (but not the RRP header) up to the application program. The message passed up from RRP to the application on host 2 is exactly the same message as the application passed down to RRP on host 1; the application does not see any of the headers that have been attached to it to implement the lower-level communication services. This whole process is illustrated in . Note that in this example, nodes in the network (e.g., switches and routers) may inspect the HHP header at the front of the message.
Note that when we say a low-level protocol does not interpret the message it is given by some high-level protocol, we mean that it does not know how to extract any meaning from the data contained in the message. It is sometimes the case, however, that the low-level protocol applies some simple transformation to the data it is given, such as to compress or encrypt it. In this case, the protocol is transforming the entire body of the message, including both the original application's data and all the headers attached to that data by higher-level protocols.
Multiplexing and Demultiplexing
Recall from Section 1.2.2 that a fundamental idea of packet switching is to multiplex multiple flows of data over a single physical link. This same idea applies up and down the protocol graph, not just to switching nodes. In , for example, we can think of RRP as implementing a logical communication channel, with messages from two different applications multiplexed over this channel at the source host and then demultiplexed back to the appropriate application at the destination host.
Practically speaking, all this means is that the header that RRP attaches to its messages contains an identifier that records the application to which the message belongs. We call this identifier RRP's demultiplexing key, or demux key for short. At the source host, RRP includes the appropriate demux key in its header. When the message is delivered to RRP on the destination host, it strips its header, examines the demux key, and demultiplexes the message to the correct application.
RRP is not unique in its support for multiplexing; nearly every protocol implements this mechanism. For example, HHP has its own demux key to determine which messages to pass up to RRP and which to pass up to MSP. However, there is no uniform agreement among protocols—even those within a single network architecture—on exactly what constitutes a demux key. Some protocols use an 8-bit field (meaning they can support only 256 high-level protocols), and others use 16-or 32-bit fields. Also, some protocols have a single demultiplexing field in their header, while others have a pair of demultiplexing fields. In the former case, the same demux key is used on both sides of the communication, while in the latter case, each side uses a different key to identify the high-level protocol (or application program) to which the message is to be delivered.
1.3.2 OSI Architecture
The ISO was one of the first organizations to formally define a common way to connect computers. Their architecture, called the Open Systems Interconnection (OSI) architecture and illustrated in, defines a partitioning of network functionality into seven layers, where one or more protocols implement the functionality assigned to a given layer. In this sense, the schematic given in is not a protocol graph, per se, but rather a reference model for a protocol graph. The ISO, usually in conjunction with a second standards organization known as the International Telecommunications Union (ITU),1 publishes a series of protocol specifications based on the OSI architecture. This series is sometimes called the "X dot" series since the protocols are given names like X.25, X.400, X.500, and so on.
Starting at the bottom and working up, the physical layer handles the transmission of raw bits over a communications link. The data link layer then collects a stream of bits into a larger aggregate called a frame. Network adaptors, along with device drivers running in the node's OS, typically implement the data link level. This means that frames, not raw bits, are actually delivered to hosts. The network layer handles routing among nodes within a packet-switched network. At this layer, the unit of data exchanged among nodes is typically called a packet rather than a frame, although they are fundamentally the same thing. The lower three layers are implemented on all network nodes, including switches within the network and hosts connected along the exterior of the network. The transport layer then implements what we have up to this point been calling a process-to- process channel. Here, the unit of data exchanged is commonly called a message rather than a packet or a frame. The transport layer and higher layers typically run only on the end hosts and not on the intermediate switches or routers.
There is less agreement about the definition of the top three layers. Skipping ahead to the top (seventh) layer, we find the application layer. Application layer protocols include things like the File Transfer Protocol (FTP), which defines a protocol by which file transfer applications can interoperate. Below that, the presentation layer is concerned with the format of data exchanged between peers, for example, whether an integer is 16, 32, or 64 bits long and whether the most significant byte is transmitted first or last, or how a video stream is formatted. Finally, the session layer provides a name space that is used to tie together the potentially different transport streams that are part of a single application. For example, it might manage an audio stream and a video stream that are being combined in a teleconferencing application.
1.3.3 Internet Architecture
The Internet architecture, which is also sometimes called the TCP/IP architecture after its two main protocols, is depicted in . An alternative representation is given in . The Internet architecture evolved out of experiences with an earlier packet-switched network called the ARPANET. Both the Internet and the ARPANET were funded by the Advanced Research Projects Agency (ARPA), one of the R&D funding agencies of the U.S. Department of Defense. The Internet and ARPANET were around before the OSI architecture, and the experience gained from building them was a major influence on the OSI reference model.
While the seven-layer OSI model can, with some imagination, be applied to the Internet, a four-layer model is often used instead. At the lowest level are a wide variety of network protocols, denoted NET1,NET2, and so on. In practice, these protocols are implemented by a combination of hardware (e.g., a network adaptor) and software (e.g., a network device driver). For example, you might find Ethernet or Fiber Distributed Data Interface (FDDI) protocols at this layer. (These protocols in turn may actually involve several sublayers, but the Internet architecture does not presume anything about them.) The second layer consists of a single protocol—the Internet Protocol (IP). This is the protocol that supports the interconnection of multiple networking technologies into a single, logical internetwork. The third layer contains two main protocols—the Transmission Control Protocol (TCP) and the User Datagram Protocol (UDP). TCP and UDP provide alternative logical channels to application programs: TCP provides a reliable byte-stream channel, and UDP provides an unreliable datagram delivery channel (datagram may be thought of as a synonym for message). In the language of the Internet, TCP and UDP are sometimes called end-to-end protocols, although it is equally correct to refer to them as transport protocols.
Running above the transport layer are a range of application protocols, such as FTP, TFTP (Trivial File Transport Protocol), Telnet (remote login), and SMTP (Simple Mail Transfer Protocol, or electronic mail), that enable the interoperation of popular applications. To understand the difference between an application layer protocol and an application, think of all the different World Wide Web browsers that are available (Firefox, Safari, Internet Explorer, Lynx, etc.). There is a similarly large number of different implementations of web servers. The reason that you can use any one of these application programs to access a particular site on the Web is because they all conform to the same application layer protocol: HTTP (HyperText Transport Protocol). Confusingly, the same word sometimes applies to both an application and the application layer protocol that it uses (e.g., FTP).
The Internet architecture has three features that are worth highlighting. First, as best illustrated by , the Internet architecture does not imply strict layering. The application is free to bypass the defined transport layers and to directly use IP or one of the underlying networks. In fact, programmers are free to define new channel abstractions or applications that run on top of any of the existing protocols.
Second, if you look closely at the protocol graph in , you will notice an hourglass shape—wide at the top, narrow in the middle, and wide at the bottom. This shape actually reflects the central philosophy of the architecture. That is, IP serves as the focal point for the architecture—it defines a common method for exchanging packets among a wide collection of networks. Above IP can be arbitrarily many transport protocols, each offering a different channel abstraction to application programs. Thus, the issue of delivering messages from host to host is completely separated from the issue of providing a useful process-to-process communication service. Below IP, the architecture allows for arbitrarily many different network technologies, ranging from Ethernet to wireless to single point-to-point links.
A final attribute of the Internet architecture (or more accurately, of the IETF culture) is that in order for a new protocol to be officially included in the architecture, there needs to be both a protocol specification and at least one (and preferably two) representative implementations of the specification. The existence of working implementations is required for standards to be adopted by the IETF. This cultural assumption of the design community helps to ensure that the architecture's protocols can be efficiently implemented. Perhaps the value the Internet culture places on working software is best exemplified by a quote on T-shirts commonly worn at IETF meetings:
We reject kings, presidents, and voting. We believe in rough consensus and running code. (Dave Clark)
Of these three attributes of the Internet architecture, the hourglass design philosophy is important enough to bear repeating. The hourglass's narrow waist represents a minimal and carefully chosen set of global capabilities that allows both higher-level applications and lower-level communication technologies to coexist, share capabilities, and evolve rapidly. The narrow-waisted model is critical to the Internet's ability to adapt rapidly to new user demands and changing technologies.
1.4 IMPLEMENTING NETWORK SOFTWARE
Network architectures and protocol specifications are essential things, but a good blueprint is not enough to explain the phenomenal success of the Internet: The number of computers connected to the Internet has roughly doubled every 12 to 18 months since 1981, and is now estimated at 350 million; the number of people that use the Internet is estimated at 1 billion; and it is believed that the number of bits transmitted over the Internet, which has also grown exponentially, surpassed the corresponding figure for the voice phone system sometime in 2001.
What explains the success of the Internet? There are certainly many contributing factors (includi
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