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Networking Basics



TCP is a connection-oriented transport protocol that sends data as an unstructured stream of bytes. By using sequence numbers and acknowledgment messages, TCP can provide a sending node with delivery information about packets transmitted to a destination node. Where data has been lost in transit from source to destination, TCP can retransmit the data until either a timeout condition is reached or until successful delivery has been achieved. TCP can also recognize duplicate messages and will discard them appropriately. If the sending computer is transmitting too fast for the receiving computer, TCP can employ flow control mechanisms to slow data transfer. TCP can also communicate delivery information to the upper-layer protocols and applications it supports.

TCP provides the following facilities to:

Stream Data Transfer
From the application's viewpoint, TCP transfers a contiguous stream of bytes. TCP does this by grouping the bytes in TCP segments, which are passed to IP for transmission to the destination. TCP itself decides how to segment the data and it may forward the data at its own convenience.
TCP assigns a sequence number to each byte transmitted, and expects a positive acknowledgment (ACK) from the receiving TCP. If the ACK is not received within a timeout interval, the data is retransmitted. The receiving TCP uses the sequence numbers to rearrange the segments when they arrive out of order, and to eliminate duplicate segments.
Flow Control
The receiving TCP, when sending an ACK back to the sender, also indicates to the sender the number of bytes it can receive beyond the last received TCP segment, without causing overrun and overflow in its internal buffers. This is sent in the ACK in the form of the highest sequence number it can receive without problems.
To allow for many processes within a single host to use TCP communication facilities simultaneously, the TCP provides a set of addresses or ports within each host. Concatenated with the network and host addresses from the internet communication layer, this forms a socket. A pair of sockets uniquely identifies each connection.
Logical Connections
The reliability and flow control mechanisms described above require that TCP initializes and maintains certain status information for each data stream. The combination of this status, including sockets, sequence numbers and window sizes, is called a logical connection. Each connection is uniquely identified by the pair of sockets used by the sending and receiving processes.

Full Duplex
TCP provides for concurrent data streams in both directions.


The User Datagram Protocol (UDP) supports network applications that need to transport data between computers. Applications that use UDP include client/server programs like video conferencing systems. Although UDP has been in use for many years -- and overshadowed by more glamorous alternatives -- it remains an interesting and viable technology.

In general, UDP implements a fairly "lightweight" layer above the Internet Protocol. UDP's main purpose is to abstract network traffic in the form of datagrams. A datagram comprises one single "unit" of binary data; the first eight (8) bytes of a datagram contain the header information and the remaining bytes contain the data itself.

UDP Headers
The UDP header consists of four (4) fields of two bytes each:
  • source port number
  • destination port number
  • datagram size
  • checksum
UDP port numbers allow different applications to maintain their own "channels" for data; both UDP and TCP use this mechanism to support multiple applications sending and receiving data concurrently. The sending application (that could be a client or a server) sends UDP datagrams through the source port, and the recipient of the packet accepts this datagram through the destination port. Some applications use static port numbers that are reserved for or registered to the application. Other applications use dynamic (unregistered) port numbers. Because the UDP port headers are two bytes long, valid port numbers range from 0 to 65535; by convention, values above 49151 represent dynamic ports.

The UDP datagram size is a simple count of the number of bytes contained in the header and data sections . Because the header length is a fixed size, this field essentially refers to the length of the variable-sized data portion (sometimes called the payload). The maximum size of a datagram varies depending on the operating environment. With a two-byte size field, the theoretical maximum size is 65535 bytes. However, some implementations of UDP restrict the datagram to a smaller number -- sometimes as low as 8192 bytes.

UDP checksums work as a safety feature. The checksum value represents an encoding of the datagram data that is calculated first by the sender and later by the receiver. Should an individual datagram be tampered with (due to a hacker) or get corrupted during transmission (due to line noise, for example), the calculations of the sender and receiver will not match, and the UDP protocol will detect this error. The algorithm is not fool-proof, but it is effective in many cases. In UDP, checksumming is optional -- turning it off squeezes a little extra performance from the system -- as opposed to TCP where checksums are mandatory.


IP is the primary layer 3 protocol in the Internet suite. In addition to internetwork routing, IP provides error reporting and fragmentation and reassembly of information units called datagrams for transmission over networks with different maximum data unit sizes. IP represents the heart of the Internet protocol suite.

IP addresses are globally unique, 32-bit numbers assigned by the Network Information Center. Globally unique addresses permit IP networks anywhere in the world to communicate with each other.

An IP address is divided into three parts. The first part designates the network address, the second part designates the subnet address, and the third part designates the host address.

IP addressing supports three different network classes. Class A networks are intended mainly for use with a few very large networks, because they provide only 8 bits for the network address field. Class B networks allocate 16 bits, and Class C networks allocate 24 bits for the network address field. Class C networks only provide 8 bits for the host field, however, so the number of hosts per network may be a limiting factor. In all three cases, the leftmost bit(s) indicate the network class. IP addresses are written in dotted decimal format; for example, Figure 2 shows the address formats for Class A, B, and C IP networks.

IP networks also can be divided into smaller units called subnetworks or "subnets." Subnets provide extra flexibility for the network administrator. For example, assume that a network has been assigned a Class A address and all the nodes on the network use a Class A address. Further assume that the dotted decimal representation of this network's address is (All zeros in the host field of an address specify the entire network.) The administrator can subdivide the network using subnetting. This is done by "borrowing" bits from the host portion of the address and using them as a subnet field, as depicted in Figure 3.

If the network administrator has chosen to use 8 bits of subnetting, the second octet of a Class A IP address provides the subnet number. In our example, address refers to network 34, subnet 1; address refers to network 34, subnet 2, and so on.

The number of bits that can be borrowed for the subnet address varies. To specify how many bits are used and where they are located in the host field, IP provides subnet masks. Subnet masks use the same format and representation technique as IP addresses. Subnet masks have ones in all bits except those that specify the host field. For example, the subnet mask that specifies 8 bits of subnetting for Class A address is The subnet mask that specifies 16 bits of subnetting for Class A address is Both of these subnet masks are pictured in Figure 4. Subnet masks can be passed through a network on demand so that new nodes can learn how many bits of subnetting are being used on their network.

Traditionally, all subnets of the same network number used the same subnet mask. In other words, a network manager would choose an eight-bit mask for all subnets in the network. This strategy is easy to manage for both network administrators and routing protocols. However, this practice wastes address space in some networks. Some subnets have many hosts and some have only a few, but each consumes an entire subnet number. Serial lines are the most extreme example, because each has only two hosts that can be connected via a serial line subnet.

As IP subnets have grown, administrators have looked for ways to use their address space more efficiently. One of the techniques that has resulted is called Variable Length Subnet Masks (VLSM). With VLSM, a network administrator can use a long mask on networks with few hosts and a short mask on subnets with many hosts. However, this technique is more complex than making them all one size, and addresses must be assigned carefully.

Of course in order to use VLSM, a network administrator must use a routing protocol that supports it. Cisco routers support VLSM with Open Shortest Path First (OSPF), Integrated Intermediate System to Intermediate System (Integrated IS-IS), Enhanced Interior Gateway Routing Protocol (Enhanced IGRP), and static routing.

On some media, such as IEEE 802 LANs, IP addresses are dynamically discovered through the use of two other members of the Internet protocol suite: Address Resolution Protocol (ARP) and Reverse Address Resolution Protocol (RARP). ARP uses broadcast messages to determine the hardware (MAC layer) address corresponding to a particular network-layer address. ARP is sufficiently generic to allow use of IP with virtually any type of underlying media access mechanism. RARP uses broadcast messages to determine the network-layer address associated with a particular hardware address. RARP is especially important to diskless nodes, for which network-layer addresses usually are unknown at boot time.

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