A tutorial on TCP/IP


I keep seeing requests on various newsgroups for an introduction to TCP/IP. I also get such requests locally. I believe that the only appropriate description of TCP/IP is the RFC's. However I also think a brief introduction is likely to be helpful before plowing right into them. The following document is an attempt to do that. It also recommends some RFC's to look at and tells you how to get them.


This document is a brief introduction to TCP/IP, followed by advice on what to read for more information. This is not intended to be a complete description, but merely enough of an introduction to allow you to start reading the RFC's. At the end of the document there will be a list of the RFC's that we recommend reading.

TCP/IP is a set of protocols developed to allow cooperating computers to share resources across a network. It was developed by a community of researchers centered around the ARPAnet. Certainly the ARPAnet is the best-known TCP/IP network. However as of June, 87, at least 130 different vendors had products that support TCP/IP, and thousands of networks of all kinds use it.

First some basic definitions. Although TCP/IP (or IP/TCP) seems to be the most common term these days, most of the documentation refers to the "Internet protocols". The Internet is a collection of networks, including the Arpanet, NSFnet, regional networks such as NYsernet, local networks at a number of University and research institutions, and a number of military networks. The term "Internet" applies to this entire set of networks. The subset of them which is managed by the Department of Defense is referred to as the "DDN" (Defense Data Network). This includes some research-oriented networks, such as the Arpanet, as well as more strictly military ones. (Because much of the funding for Internet protocol developments is done via the DDN organization, the terms Internet and DDN can sometimes seem equivalent.) All of these networks are connected to each other, and users can send messages from any of them to any other (except where security or other policy restrictions control access). Officially speaking, the Internet protocol documents are simply standards adopted by the Internet community for its own use. More recently, the Department of Defense issued a MILSPEC definition of TCP/IP. This was intended to be a more formal definition, appropriate for use in purchasing specifications. However most of the TCP/IP community continues to use the Internet standards. The MILSPEC version is intended to be consistent with it.

Whatever it is called, TCP/IP is a family of protocols. A few are basic ones used for many applications. These include IP, TCP, and UDP. Others are protocols for doing specific tasks, e.g. transferring files between computers, sending mail, or finding out who is logged in on another computer. Any real application will use several of these protocols. A typical situation is sending mail. First, there is a protocol for mail. This defines a set of commands which one machine sends to another, e.g. commands to specify who the sender of the message is, who it is being sent to, and then the text of the message. However this protocol assumes that there is a way to communicate reliably between the two computers. Mail, like other application protocols, simply defines a set of commands and messages to be sent. It is designed to be used together with TCP and IP. TCP is responsible for making sure that the commands get through to the other end. It keeps track of what is sent, and retransmitts anything that did not get through. If any message is too large for one packet, e.g. the text of the mail, TCP will split it up into several packets, and make sure that they all arrive correctly. Since these functions are needed for many applications, they are put together into a separate protocol, rather than being part of the specifications for sending mail. You can think of TCP as forming a library of routines that applications can use when they need reliable network communications with another computer. Similarly, TCP calls on the services of IP. Although the services that TCP supplies are needed by many applications, there are still some kinds of applications that don't need them. However there are some services that every application needs. So these services are put together into IP. As with TCP, you can think of IP as a library of routines that TCP calls on, but which is also available to applications that don't use TCP. This strategy of building several levels of protocol is called "layering". We think of the applications programs such as mail, TCP, and IP, as being separate "layers", each of which calls on the services of the layer below it. Generally, TCP/IP applications use 4 layers:

  - an application protocol such as mail

  - a protocol such as TCP that provides services need by many applications

  - IP, which provides the basic service of getting packets to their

  - the protocols needed to manage a specific physical medium, such as
	Ethernet or a point to point line.
TCP/IP is based on the "catenet model". (This is described in more detail in ien-48.txt.) This model assumes that there are a large number of independent networks connected together by gateways. The user should be able to access computers or other resources on any of these networks. Packets will often pass through a dozen different networks before getting to their final destination. The routing needed to accomplish this should be completely invisible to the user. As far as the user is concerned, all he needs to know in order to access another system is an "Internet address". This is an address that looks like It is actually a 32-bit number. However it is normally written as 4 decimal numbers, each representing 8 bits of the address. (The term "octet" is used by Internet documentation for such 8-bit chunks. The term "byte" is not used, because TCP/IP is supported by some computers that have byte sizes other than 8 bits.) Generally the structure of the address gives you some information about how to get to the system. For example, 128.6 is a network number assigned by a central authority to Rutgers University. Rutgers uses the next octet to indicate which of the campus Ethernets is involved. 128.6.4 happens to be an Ethernet used by the Computer Science Department. The last octet allows for up to 254 systems on each Ethernet. Note that and would be different systems. (The structure of an Internet address is described in a bit more detail later.)

Of course we normally refer to systems by name, rather than by Internet address. When we specify a name, the network software looks it up in a database, and comes up with the corresponding Internet address. Most of the network software deals strictly in terms of the address. (rfc-882.txt describes the database used to look up names.)

TCP/IP is a "connectionless" protocol. Information is transfered in "packets". Each of these packets is sent through the network individually. There are provisions to open connections to systems. However at some level, information is put into packets, and those packets are treated by the network as completely separate. For example, suppose you want to transfer a 15000 octet file. Most networks can't handle a 15000 octet packet. So the protocols will break this up into something like 30 500-octet packets. Each of these packets will be sent to the other end. At that point, they will be put back together into the 15000-octet file. However while those packets are in transit, the network doesn't know that there is any connection between them. It is perfectly possible that packet 14 will actually arrive before packet 13. It is also possible that somewhere in the network, an error will occur, and a packet won't get through at all. In that case, that packet has to be sent again. In fact, there are two separate protocols involved in doing this. TCP (the "transmission control protocol") is responsible for breaking up the message into packets, reassembling them at the other end, resending anything that gets lost, and putting things back in the right order. IP (the "internet protocol") is responsible for routing individual packets. It may seem like TCP is doing all the work. And in small networks that is true. However in the Internet, simply getting a packet to its destination can be a complex job. A connection may require the packet to go through several networks at Rutgers, a serial line to the John von Neuman Supercomputer Center, a couple of Ethernets there, a series of 56Kbaud phone lines to another NSFnet site, and more Ethernets on another campus. Keeping track of the routes to all of the destinations and handling incompatibilities among different transport media turns out to be a complex job. Note that the interface between TCP and IP is fairly simple. TCP simply hands IP a packet with a destination. IP doesn't know how this packet relates to any packet before it or after it.

It may have occured to you that something is missing here. We have talked about Internet addresses, but not about how you keep track of multiple connections to a given system. Clearly it isn't enough to get a packet to the right destination. TCP has to know which connection this packet is part of. This task is referred to as "demultiplexing." In fact, there are several levels of demultiplexing going on in TCP/IP. The information needed to do this demultiplexing is contained in a series of "headers". A header is simply a few extra octets tacked onto the beginning of a packet by some protocol in order to keep track of it. It's a lot like putting a letter into an envelope and putting an address on the outside of the envelope. Except with modern networks it happens several times. It's like you put the letter into a little envelope, your secretary puts that into a somewhat bigger envelope, the campus mail center puts that envelope into a still bigger one, etc. Here is an overview of the headers that get stuck on a message that passes through a typical TCP/IP network:

We start with a single data stream, say a file you are trying to send to some other computer:


TCP breaks it up into managable chunks. (In order to do this, TCP has to know how large a packet your network can handle. Actually, the TCP's at each end say how big a packet they can handle, and then they pick the smallest size.)

.... .... .... .... .... .... .... .... .....

TCP puts a header at the front of each packet. This header actually contains at least 20 octets, but the most important ones are a source and destination "port number" and a "sequence number". The port numbers are used to keep track of different conversations. Suppose 3 different people are transferring files. Your TCP might allocate port numbers 1000, 1001, and 1002 to these transfers. When you are sending a packet, this becomes the "source" port number, since you are the source of the packet. Of course the TCP at the other end has assigned a port number of its own for the conversation. Your TCP has to know the port number used by the other end as well. (It finds out when the connection starts, as we will explain below.) It puts this in the "destination" port field. Of course if the other end sends a packet back to you, the source and destination port numbers will be reversed, since then it will be the source and you will be the destination. Each packet has a sequence number. This is used so that the other end can make sure that it gets the packets in the right order, and that it hasn't missed any. (See the TCP specification for details. TCP doesn't number the packets, but the octets. So if there are 500 octets of data in each packet, the first packet might be numbered 0, the second 500, the next 1000, the next 1500, etc.) Finally, I will mention the Checksum. This is a number that is computed by adding up all the octets in the packet (more or less - see the TCP spec). The result is put in the header. TCP at the other end computes the checksum again. If they disagree, then something bad happened to the packet in transmission, and it is thrown away. So here's what the packet looks like now.

   |          Source Port          |       Destination Port        |
   |                        Sequence Number                        |
   |                      various other junk                       |
   |                      various other junk                       |  
   |           Checksum            |           other junk          |
   |   your data ... next 500 octets                               |
   |   ......                                                      |
If we abbreviate the TCP header as "T", the whole file now looks like this:

T.... T.... T.... T.... T.... T.... T.... T....

TCP now sends each of these packets to IP. Of course it has to tell IP the Internet address of the computer at the other end. Note that this is all IP is concerned about. It doesn't care about what is in the packet, or even in the TCP header. IP's job is simply to find a route for the packet and get it to the other end. In order to allow gateways or other intermediate systems to forward the packet, it adds its own header. The main things in this header are the source and destination Internet address (32-bit addresses, like, the protocol number, and another checksum. The source Internet address is simply the address of your machine. (This is necessary so the other end knows where the packet came from.) The destination Internet address is the address of the other machine. (This is necessary so any gateways in the middle know where you want the packet to go.) The protocol number tells IP at the other end to send the packet to TCP. Although most IP traffic uses TCP, there are other protocols that can use IP, so you have to tell IP which protocol to send the packet to. Finally, the checksum allows IP at the other end to verify that the packet wasn't damaged in transit. Note that TCP and IP have separate checksums. This is because IP doesn't know anything about TCP. As far as IP is concerned, everything after its header is just a bunch of bits. So IP computes a checksum of its own header, and IP at the other end checks it to make sure that the message didn't get damaged in transit. Once IP has tacked on its header, here's what the message looks like:

   |                      various other junk                       |  
   |                      various other junk                       |  
   |     junk      |    Protocol   |         Header Checksum       |
   |                       Source Address                          |
   |                    Destination Address                        |
   |  TCP header, then your data ......
If we represent the IP header by an "I", your file now looks like this:

IT.... IT.... IT.... IT.... IT.... IT.... IT.... IT....

At this point, it's possible that no more headers are needed. If your computer happens to have a direct phone line connecting it to the destination computer, or to a gateway, it may simply send the packets out on the line (though likely a synchronous protocol such as HDLC would be used, and it would add at least a few octets at the beginning and end).

However most of our networks these days use Ethernet. So now we have to describe Ethernet's headers. Unfortunately, Ethernet has its own addresses. The people who designed Ethernet wanted to make sure that no two machines would end up with the same Ethernet address. Furthermore, they didn't want the user to have to worry about assigning addresses. So each Ethernet controller comes with an address builtin from the factory. In order to make sure that they would never have to reuse addresses, the Ethernet designers allocated 48 bits for the Ethernet address. People who make Ethernet equipment have to register with a central authority, to make sure that the numbers they assign don't overlap any other manufacturer. Ethernet is a "broadcast medium". That is, it is in effect like an old party line telephone. When you send a packet out on the Ethernet, every machine on the network sees the packet. So something is needed to make sure that the right machine gets it. As you might guess, this involves the Ethernet header. Every Ethernet packet has a 14-octet header that includes the source and destination Ethernet address, and a type code. Each machine is supposed to pay attention only to packets with its own Ethernet address in the destination field. (It's perfectly possible to cheat, which is one reason that Ethernet communications are not terribly secure.) Note that there is no connection between the Ethernet address and the Internet address. Each machine has to have a table of what Ethernet address corresponds to what Internet address. (We will describe how this table is constructed a bit later.) In addition to the addresses, the header contains a type code. The type code is to allow for several different protocol families to be used on the same network. So you can use TCP/IP, DECnet, Xerox NS, etc. at the same time. Each of them will put a different value in the type field. Finally, there is a checksum. The Ethernet controller computes a checksum of the entire packet. When the other end receives the packet, it recomputes the checksum, and throws the packet away if the answer disagrees with the original. The checksum is put on the end of the packet, not in the header. The final result is that your message looks like this:

   |       Ethernet destination address (first 32 bits)            |
   | Ethernet dest (last 16 bits)  |Ethernet source (first 16 bits)|
   |       Ethernet source address (last 32 bits)                  |
   |        Type code              |
   |  IP header, then TCP header, then your data                   |
   |                                                               |
   |                                                               |
   |   end of your data                                            |
   |                       Ethernet Checksum                       |
If we represent the Ethernet header with "E", and the Ethernet checksum with "C", your file now looks like this: EIT....C EIT....C EIT....C EIT....C EIT....C EIT....C

When these packets are received by the other end, of course all the headers are removed. The Ethernet interface removes the Ethernet header and the checksum. It looks at the type code. Since the type code is the one assigned to IP, the Ethernet device driver passes the packet up to IP. IP removes the IP header. It looks at the IP protocol field. Since the protocol type is TCP, it passes the packet up to TCP. TCP now looks at the packet sequence number. It uses the sequence numbers and other information to combine all the packets into the original file.

The ends our initial summary of TCP/IP. There are still some crucial concepts we haven't gotten to, so we'll now go back and add details in several areas. (For detailed descriptions of the items discussed here see, rfc793.txt for TCP, rfc791.txt for IP, and rfc894.txt and rfc826.txt for sending IP over Ethernet.)

Well-known sockets and the applications layer
So far, we have described how a stream of data is broken up into packets, sent to another computer, and put back together. However something more is needed in order to accomplish anything useful. There has to be a way for you to open a connection to a specified computer, log into it, tell it what file you want, and control the transmission of the file. (If you have a different application in mind, e.g. computer mail, some analogous protocol is needed.) This is done by "application protocols". The application protocols run "on top" of TCP/IP. That is, when they want to send a message, they give the message to TCP. TCP makes sure it gets delivered to the other end. Because TCP and IP take care of all the networking details, the applications protocols can treat a network connection as if it were a simple byte stream, like a terminal or phone line.

Before going into more details about applications programs, we have to describe how you find an application. Suppose you want to send a file to a computer whose Internet address is To start the process, you need more than just the Internet address. You have to connect to the file transfer server at the other end. In general, network programs are specialized for a specific set of tasks. Most systems have separate programs to handle file transfers, remote terminal logins, mail, etc. When you connect to, you have to specify that you want to talk to the file transfer program. This is done by having "well-known sockets" for each program. Recall that TCP uses port numbers to keep track of individual conversations. User programs normally use more or less random port numbers. However specific port numbers are assigned to the programs that sit waiting for requests. For example, if you want to send a file, you will start a program called "ftp". It will open a connection using some random number, say 1234, for the port number on its end. However it will specify port number 21 for the other end. This is the official port number for the ftp server. Note that there are two different programs involved. You run ftp on your side. This is a program designed to accept commands from your terminal and pass them on to the other end. The program that you talk to on the other machine is the ftp server. It is designed to accept commands from the network connection, rather than an interactive terminal. There is no need for your program to use a well-known socket number for itself. Nobody is trying to find it. However the servers have to have well-known numbers, so that people can open connections to them and start sending them commands. The official port numbers for each program are given in "Assigned Numbers".

Note that a connection is actually described by a set of 4 numbers: the Internet address at each end, and the TCP port number at each end. Every packet has all four of those numbers in it. (The Internet addresses are in the IP header, and the TCP port numbers are in the TCP header.) In order to keep things straight, no two connections can have the same set of numbers. However it is enough for any one number to be different. For example, it is perfectly possible for two different users on a machine to be sending files to the same other machine. This could result in connections with the following parameters:

               Internet addresses         TCP ports
connection 1,      1234, 21
connection 2,      1235, 21
Since the same machines are involved, the Internet addresses are the same. Since they are both doing file transfers, one end of the connection involves the well-known port number for file transfer. The only thing that differs is the port number for the program that the users are running. That's enough of a difference. Generally, at least one end of the connection asks the network software to assign it a port number that is guaranteed to be unique. Normally, it's the user's end, since the server has to use a well-known number.

Now that we know how to open connections, let's get back to the applications programs. As mentioned above, once TCP has opened a connection, we have something that might as well be a simple wire. All the hard parts are handled by TCP and IP. However we still need some agreement as to what we send over this connection. In effect this is simply an agreement on what set of commands the application will understand, and the format in which they are to be sent. Generally, what is sent is a combination of commands and data. They use context to differentiate. For example, the mail protocol works like this: Your mail program opens a connection to the mail server at the other end. Your program gives it your machine's name, the sender of the message, and the recipients you want it sent to. It then sends a command saying that it is starting the message. At that point, the other end stops treating what it sees as commands, and starts accepting the message. Your end then starts sending the text of the message. At the end of the message, a special mark is sent (a dot in the first column). After that, both ends understand that your program is again sending commands. This is the simplest way to do things, and the one that most applications use.

File transfer is somewhat more complex. The file transfer protocol involves two different connections. It starts out just like mail. The user's program sends commands like "log me in as this user", "here is my password", "send me the file with this name". However once the command to send data is sent, a second connection is opened for the data itself. It would certainly be possible to send the data on the same connection, as mail does. However file transfers often take a long time. The designers of the file transfer protocol wanted to allow the user to continue issuing commands while the transfer is going on. For example, the user might make an inquiry, or he might abort the transfer. Thus the designers felt it was best to use a separate connection for the data and leave the original command connection for commands. (It is also possible to open command connections to two different computers, and tell them to send a file from one to the other. In that case, the data couldn't go over the command connection.)

Remote terminal connections use another mechanism still. For remote logins, there is just one connection. It normally sends data. When it is necessary to send a command (e.g. to set the terminal type or to change some mode), a special character is used to indicate that the next character is a command. If the user happens to type that special character as data, two of them are sent.

We are not going to describe the application protocols in detail in this document. It's better to read the RFC's yourself. However there are a couple of common conventions used by applications that will be described here. First, the common network representation: TCP/IP is intended to be usable on any computer. Unfortunately, not all computers agree on how data is represented. There are differences in character codes (ASCII vs. EBCDIC), in end of line conventions (carriage return, line feed, or a representation using counts), and in whether terminals expect characters to be sent individually or a line at a time. In order to allow computers of different kinds to communicate, each applications protocol defines a standard representation. Note that TCP and IP do not care about the representation. TCP simply sends octets. However the programs at both ends have to agree on how the octets are to be interpreted. The RFC for each application specifies the standard representation for that application. Normally it is "net ASCII". This uses ASCII characters, with end of line denoted by a carriage return followed by a line feed. For remote login, there is also a definition of a "standard terminal", which turns out to be a half-duplex terminal with echoing happening on the local machine. Most applications also make provisions for the two computers to agree on other representations that they may find more convenient. For example, PDP-10's have 36-bit words. There is a way that two PDP-10's can agree to send a 36-bit binary file. Similarly, two systems that prefer full-duplex terminal conversations can agree on that. However each application has a standard representation, which every machine must support.

(For more details about the protocols mentioned in this section, see rfc821.txt and rfc822.txt for mail, rfc959.txt for file transfer, and rfc854.txt and rfc855.txt for remote logins. For the well-known port numbers, see the current edition of Assigned Numbers, and possible rfc814.txt.)

Protocols other than TCP: UDP and ICMP
So far, we have described only connections that use TCP. Recall that TCP is responsible for breaking up messages into packets, and reassembling them properly. However in many applications, we have messages that will always fit in a single packet. An example is name lookup. When a user attempts to make a connection to another system, he will generally specify the system by name, rather than Internet address. His system has to translate that name to an address before it can do anything. Generally, only a few systems have the database used to translate names to addresses. So the user's system will want to send a query to one of the systems that has the database. This query is going to be very short. It will certainly fit in one packet. So will the answer. Thus it seems silly to use TCP. Of course TCP does more than just break things up into packets. It also makes sure that the data arrives, resending packets where necessary. But for a question that fits in a single packet, we don't need all the complexity of TCP to do this. If we don't get an answer after a few seconds, we can just ask again. For applications like this, there are alternatives to TCP.

The most common alternative is UDP ("user datagram protocol"). UDP is designed for applications where you don't need to put sequences of packets together. It fits into the system much like TCP. There is a UDP header. The network software puts the UDP header on the front of your data, just as it would put a TCP header on the front of your data. Then UDP sends the data to IP, which adds the IP header, putting UDP's protocol number in the protocol field instead of TCP's protocol number. However UDP doesn't do as much as TCP does. It doesn't split data into multiple packets. It doesn't keep track of what it has sent so it can resend if necessary. About all that UDP provides is port numbers, so that several programs can use UDP at once. UDP port numbers are used just like TCP port numbers. There are well-known port numbers for servers that use UDP. Note that the UDP header is shorter than a TCP header. It still has source and destination port numbers, and a checksum, but that's about it. No sequence number, since it is not needed. UDP is used by the protocols that handle name lookups (see ien-116.txt, rfc882.txt, and rfc883.txt), and a number of similar protocols.

Another alternative protocol is ICMP ("Internet control message protocol"). ICMP is used for error messages, and other messages intended for the TCP/IP software itself, rather than any particular user program. For example, if you attempt to connect to a host, your system may get back an ICMP message saying "host unreachable". ICMP can also be used to find out some information about the network. See rfc792.txt for details of ICMP. ICMP is similar to UDP, in that it handles messages that fit in one packet. However it is even simpler than UDP. It doesn't even have port numbers in its header. Since all ICMP messages are interpreted by the network software itself, no port numbers are needed to say where a ICMP message is supposed to go.

The description above indicated that the IP implementation is responsible for getting packets to the destination indicated by the destination address, but little was said about how this would be done. The task of finding how to get a packet to its destination is referred to as "routing". In fact many of the details depend upon the particular implementation. However some general things can be said.

First, it is necessary to understand the model on which IP is based. IP assumes that a system is attached to some local network. We assume that the system can send packets to any other system on its own network. (In the case of Ethernet, it simply finds the Ethernet address of the destination system, and puts the packet out on the Ethernet.) The problem comes when a system is asked to send a packet to a system on a different network. This problem is handled by gateways. A gateway is a system that connects a network with one or more other networks. Gateways are often normal computers that happen to have more than one network interface. For example, we have a Unix machine that has two different Ethernet interfaces. Thus it is connected to networks 128.6.4 and 128.6.3. This machine can act as a gateway between those two networks. The software on that machine must be set up so that it will forward packets from one network to the other. That is, if a machine on network 128.6.4 sends a packet to the gateway, and the packet is addressed to a machine on network 128.6.3, the gateway will forward the packet to the destination. Major communications centers often have gateways that connect a number of different networks.

Routing in IP is based entirely upon the network number of the destination address. Each computer has a table of network numbers. For each network number, a gateway is listed. This is the gateway to be used to get to that network. Note that the gateway doesn't have to connect directly to the network. It just has to be the best place to go to get there. For example at Rutgers, our interface to NSFnet is at the John von Neuman Supercomputer Center (JvNC). Our connection to JvNC is via a high-speed serial line connected to a gateway whose address is Systems on net 128.6.3 will list as the gateway for many off-campus networks. However systems on net 128.6.4 will list as the gateway to those same off-campus networks. is the gateway between networks 128.6.4 and 128.6.3, so it is the first step in getting to JNC.

When a computer wants to send a packet, it first checks to see if the destination address is on the system's own local network. If so, the packet can be sent directly. Otherwise, the system expects to find an entry for the network that the destination address is on. The packet is sent to the gateway listed in that entry. This table can get quite big. For example, the Internet now includes several hundred individual networks. Thus various strategies have been developed to reduce the size of the routing table. One strategy is to depend upon "default routes". Often, there is only one gateway out of a network. This gateway might connect a local Ethernet to a campus-wide backbone network. In that case, we don't need to have a separate entry for every network in the world. We simply define that gateway as a "default". When no specific route is found for a packet, the packet is sent to the default gateway. A default gateway can even be used when there are several gateways on a network. There are provisions for gateways to send a message saying "I'm not the best gateway -- use this one instead." (The message is sent via ICMP. See rfc792.txt) Most network software is designed to use these messages to add entries to their routing tables. Suppose network 128.6.4 has two gateways, and leads to several other internal Rutgers networks. leads indirectly to the NSFnet. Suppose we set as a default gateway, and have no other routing table entries. Now what happens when we need to send a packet to MIT? MIT is network 18. Since we have no entry for network 18, the packet will be sent to the default, As it happens, this gateway is the wrong one. So it will forward the packet to But it will also send back an error saying in effect: "to get to network 18, use". Our software will then add an entry to the routing table. Any future packets to MIT will then go directly to

Most IP experts recommend that individual computers should not try to keep track of the entire network. Instead, they should start with default gateways, and let the gateways tell them the routes, as just described. However this doesn't say how the gateways should find out about the routes. The gateways can't depend upon this strategy. They have to have fairly complete routing tables. (It is possible to do hierarchical routing, where all of the gateways on a campus know about the campus network, but direct all off-campus traffic to a single gateway with connections off-campus.) For this, some sort of routing protocol is needed. A routing protocol is simply a technique for the gateways to find each other, and keep up to date about the best way to get to every network. rfc1009.txt contains a review of gateway design and routing. However rip.doc is probably a better introduction to the subject. It contains some tutorial material, and a detailed description of the most commonly-used routing protocol.

Details about Internet addresses: subnets and broadcasting
As indicated above, Internet addresses are 32-bit numbers, normally written as 4 octets (in decimal), e.g. There are actually 3 different types of address. The problem is that the address has to indicate both the network and the host within the network. It was felt that eventually there would be lots of networks. Many of them would be small, but probably 24 bits would be needed to represent all the IP networks. It was also felt that some very big networks might need 24 bits to represent all of their hosts. This would seem to lead to 48 bit addresses. But the designers really wanted to use 32 bit addresses. So they adopted a kludge. The assumption is that most of the networks will be small. So they set up three different ranges of address. Addresses beginning with 1 to 126 use only the first octet for the network number. The other three octets are available for the host number. Thus 24 bits are available for hosts. These numbers are used for large networks. But there can only be 126 of these very big networks. The Arpanet is one, and there are a few large commercial networks. But few normal organizations get one of these "class A" addresses. For normal large organizations, "class B" addresses are used. Class B addresses use the first two octets for the network number. Thus network numbers are 128.1 through 191.254. (We avoid 0 and 255, for reasons that we see below. We also avoid addresses beginning with 127, because that is used by some systems for special purposes.) The last two octets are available for host addesses, giving 16 bits of host address. This allows for 64516 computers, which should be enough for most organizations. (It is possible to get more than one class B address, if you run out.) Finally, class C addresses use three octets, in the range 192.1.1 to 223.254.254. These allow only 254 hosts on each network, but there can be lots of these networks. Addresses above 223 are reserved for future use, as class D and E (which are currently not defined).

Many large organizations find it convenient to divide their network number into "subnet". For example, Rutgers has been assigned a class B address, 128.6. We find it convenient to use the third octet of the address to indicate which Ethernet a host is on. This division has no significance outside of Rutgers. A computer at another institution would send any packet whose destination address began with 128.6 on the best route to Rutgers. They would not have different routes for 128.6.4 or 128.6.5. But inside Rutgers, we treat 128.6.4 and 128.6.5 as separate networks. In effect, gateways inside Rutgers have separate entries for each Rutgers subnet, whereas gateways outside Rutgers just have one entry for 128.6. Note that we could do exactly the same thing by using a separate class C address for each Ethernet. As far as Rutgers is concerned, it would be just as convenient for us to have a number of class C addresses. However using class C addresses would make things inconvenient for the rest of the world. Every institution that wanted to talk to us would have to have a separate entry for each one of our networks. If every institution did this, there would be far too many networks for any reasonable gateway to keep track of. By subdividing a class B network, we hide our internal structure from everyone else, and save them trouble. This subnet strategy requires special provisions in the network software. It is described in rfc950.txt.

0 and 255 have special meanings. 0 is reserved for machines that don't know their address. In certain circumstances it is possible for a machine not to know the number of the network it is on, or even its own host address. So would be a machine that knew it was host number 23, but didn't know on what network.

255 is used for "broadcast". A broadcast is a message that you want every system on the network to see. Broadcasts are used in some situations where you don't know who to talk to. For example, suppose you need to look up a host name and get its Internet address. Sometimes you don't know the address of the system that has the host name data base. In that case, you might send the request as a broadcast. There are also cases where a number of systems are interested in information. It is then less expensive to send a single broadcast than to send packets individually to each host that is interested in the information. In order to send a broadcast, you use an address that is made by using your network address, with all ones in the part of the address where the host number goes. For example, if you are on network 128.6.4, you would use for broadcasts. How this is actually implemented depends upon the medium. It is not possible to send broadcasts on the Arpanet, or on point to point lines. However it is possible on an Ethernet. If you use an Ethernet address with all its bits on (all ones), every machine on the Ethernet is supposed to look at that packet.

Although the official broadcast address for network 128.6.4 is now, there are some other addresses that may be treated as broadcasts by certain implementations. For convenience, the standard also allows to be used. This refers to all hosts on the local network. It is often simpler to use instead of finding out the network number for the local network and forming a broadcast address such as In addition, certain older address, e.g. Finally, certain older implementations may not understand about subnets. Thus they consider the network number to be 128.6. In that case, they will assume a broadcast address of or Until support for broadcasts is implemented properly, it can be a somewhat dangerous feature to use.

Because 0 and 255 are used for unknown and broadcast addresses, normal hosts should never be given addresses containing 0 or 255. Addresses should never begin with 0, 127, or any number above 223. Addresses violating these rules are sometimes referred to as "Martians", because of rumors that the Central University of Mars is using network 225.

Packet splitting and reassembly
TCP/IP is designed for use with many different kinds of network. Unfortunately, network designers do not agree about how big packets can be. Ethernet packets can be 1500 octets long. Arpanet packets have a maximum of around 1000 octets. Some very fast networks have much larger packet sizes. At first, you might think that IP should simply settle on the smallest possible size. Unfortunately, this would cause serious performance problems. When transferring large files, big packets are far more efficient than small ones. So we want to be able to use the largest packet size possible. But we also want to be able to handle networks with small limits. There are two provisions for this. First, TCP has the ability to "negotiate" about packet size. When a TCP connection first opens, both ends can send the maximum packet size they can handle. The smaller of these numbers is used for the rest of the connection. This allows two implementations that can handle big packets to use them, but also lets them talk to implementations that can't handle them. However this doesn't completely solve the problem. The most serious problem is that the two ends don't necessarily know about all of the steps in between. For example, when sending data between Rutgers and Berkeley, it is likely that both computers will be on Ethernets. Thus they will both be prepared to handle 1500-octet packets. However the connection will at some point end up going over the Arpanet. It can't handle packets of that size. For this reason, there are provisions to split packets up into pieces. The IP header contains fields indicating the a packet has been split, and enough information to let the pieces be put back together. If a gateway connects an Ethernet to the Arpanet, it must be prepared to take 1500-octet Ethernet packets and split them into pieces that will fit on the Arpanet. Furthermore, every implementation of TCP/IP must be prepared to accept pieces and put them back together. This is referred to as "reassembly".

TCP/IP implementations differ in the approach they take to deciding on packet size. It is fairly common for implementations to use 576-byte packets whenever they can't verify that the entire path is able to handle larger packets. The problem is that many implementations have bugs in the code to reassemble pieces. So many implementors try to avoid ever having splits occur. Different implementors take different approaches to deciding when it is safe to use large packets. Some use them only for the local network. Others will use them for any network on the same campus. 576 bytes is a "safe" size, which every implementation must support.

Ethernet encapsulation: ARP
There was a brief discussion above about what IP packets looked like on an Ethernet. The discussion showed the Ethernet header and checksum. However it left one hole: It didn't say how to figure out what Ethernet address to use when you want to talk to a given Internet address. In fact, there is a separate protocol for this, called ARP ("address resolution protocol"). Note by the way that ARP is not an IP protocol. That is, the ARP packets do not have IP headers. Suppose you are on system and you want to connect to system Your system will first verify that is on the same network, so it can talk directly via Ethernet. Then it will look up in its ARP table, to see if it already knows the Ethernet address. If so, it will stick on an Ethernet header, and send the packet. But suppose this system is not in the ARP table. There is no way to send the packet, because you need the Ethernet address. So it uses the ARP protocol to send an ARP request. Essentially an ARP request says "I need the Ethernet address for". Every system listens to ARP requests. When a system sees an ARP request for itself, it is required to respond. So will see the request, and will respond with an ARP reply saying in effect " is 8:0:20:1:56:34". (Recall that Ethernet addresses are 48 bits. This is 6 octets. Ethernet addresses are conventionally shown in hex, using the punctuation shown.) Your system will save this information in its ARP table, so future packets will go directly. Most systems treat the ARP table as a cache, and clear entries in it if they have not been used in a certain period of time.

Note by the way that ARP requests must be sent as "broadcasts". There is no way that an ARP request can be sent to the right system. After all, the whole reason for sending an ARP request is that you don't know the Ethernet address. So an Ethernet address of all ones is used, i.e. ff:ff:ff:ff:ff:ff. By convention, every machine on the Ethernet is required to pay attention to packets with this as an address. So every system sees every ARP requests. They all look to see whether the request is for their own address. If so, they respond. If not, they could just ignore it. (Some hosts will use ARP requests to update their knowledge about other hosts on the network, even if the request isn't for them.) Note that packets whose IP address indicates broadcast (e.g. or are also sent with an Ethernet address that is all ones.

Getting more information
This directory contains documents describing the major protocols. There are literally hundreds of documents, so we have chosen the ones that seem most important. Internet standards are called RFC's. RFC stands for Request for Comment. A proposed standard is initially issued as a proposal, and given an RFC number. When it is finally accepted, it is added to Official Internet Protocols, but it is still referred to by the RFC number. We have also included two IEN's. (IEN's are an older form of RFC.) The convention is that whenever an RFC is revised, the revised version gets a new number. This is fine for most purposes, but it causes problems with two documents: Assigned Numbers and Official Internet Protocols. These documents are being revised all the time, so the RFC number keeps changing. You will have to look in rfc-index.txt to find the number of the latest edition. Anyone who is seriously interested in TCP/IP should read the RFC describing IP (791). RFC 1009 is also useful. It is a specification for gateways to be used by NSFnet. As such, it contains an overview of a lot of the TCP/IP technology. You should probably also read the description of at least one of the application protocols, just to get a feel for the way things work. Mail is probably a good one (821/822). TCP (793) is of course a very basic specification. However the spec is fairly complex, so you should only read this when you have the time and patience to think about it carefully. Fortunately, the author of the major RFC's (Jon Postel) is a very good writer. The TCP RFC is far easier to read than you would expect, given the complexity of what it is describing. You can look at the other RFC's as you become curious about their subject matter.

Here is a list of the documents you are more likely to want:

   rfc-index - list of all RFC's
   rfc1012  -  somewhat fuller list of all RFC's
   rfc1011  -  Official Protocols.  It's useful to scan this to
	see what tasks protocols have been built for.  This defines
	which RFC's are actual standards, as opposed to requests
	for comments.
   rfc1010  -  Assigned Numbers.  If you are working with TCP/IP,
	you will probably want a hardcopy of this as a reference.
	It's not very exciting to read.  It lists all the offically
	defined well-known ports and lots of other things.
   rfc1009  -  NSFnet gateway specifications.  A good overview of
	IP routing and gateway technology.
   rfc973   -  update on domains
   rfc959   -  FTP (file transfer)
   rfc950   -  subnets
   rfc894   -  how IP is to be put on Ethernet, see also rfc825
   rfc882/3 -  domains (the database used to go from host names to
	Internet address and back -- also used to handle UUCP
	these days).  See also rfc973
   rfc854/5 -  telnet - protocol for remote logins
   rfc826   -  ARP - protocol for finding out Ethernet addresses
   rfc821/2 -  mail
   rfc814   -  names and ports - general concepts behind well-known ports
   rfc793   -  TCP
   rfc792   -  ICMP
   rfc791   -  IP
   rfc768   -  UDP
   rip.doc  -  details of the most commonly-used routing protocol
   ien-116  -  old name server (still needed by several kinds of system)
   ien-48   -  the Catenet model, general description of the philosophy
	behind TCP/IP
The following documents are somewhat more specialized.

   rfc813   -  window and acknowledgement strategies in TCP
   rfc815   -  packet reassembly techniques
   rfc816   -  fault isolation and resolution techniques
   rfc817   -  modularity and efficiency in implementation
   rfc879   -  the maximum segment size option in TCP
   rfc896   -  congestion control
   rfc827,888,904,975,985  - EGP
To those of you who may be reading this document remotely instead of at Rutgers: The most important RFC's have been collected into a three-volume set, the DDN Protocol Handbook. It is available from the DDN Network Information Center, SRI International, 333 Ravenswood Avenue, Menlo Park, California 94025 (telephone: 800-235-3155). You should be able to get them via anonymous FTP from sri-nic.arpa. File names are:

  rip.doc is available by anonymous FTP from topaz.rutgers.edu, as
Sites with access to UUCP but not FTP may be able to retreive them via UUCP from UUCP host rutgers. The file names would be

Note that SRI-NIC has the entire set of RFC's and IEN's, but rutgers and topaz have only those specifically mentioned above.

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