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TCP/IP Model and protocols

Protocols are sets of rules for message formats and procedures that allow machines and application programs to exchange information. These rules must be followed by each machine involved in the communication in order for the receiving host to be able to understand the message.


Basic protocol functions include segmentation and reassembly, encapsulation, connection control, ordered delivery, flow control, error control, routing and multiplexing. Protocols are needed to enable parties to understand each other and make sense of received information. A protocol stack is a list of protocols (one protocol per layer). A network protocol architecture is a set of layers and protocols.


One major trend in any telecommunications network is to move towards IP network technologies. Satellite networks are following the same trend.


As with all other communications protocols, TCP/IP is composed of different layers.

First, application programs send messages or streams of data to one of the Internet Transport Layer Protocols, either the User Datagram Protocol (UDP) or the Transmission Control Protocol (TCP). These protocols receive the data from the application, divide it into smaller pieces called packets, add a destination address, and then pass the packets along to the next protocol layer, the Internet Network layer.

The Internet Network layer encloses the packet in an Internet Protocol (IP) datagram, puts in the datagram header and trailer, decides where to send the datagram (either directly to a destination or else to a gateway), and passes the datagram on to the Network Interface layer.

The Network Interface layer accepts IP datagrams and transmits them as frames over a specific network hardware, such as Ethernet or Token-Ring networks.

TCP/IP suite of protocols


Physical Layer

The physical layer represents the physical devices that interconnect computers. The physical layer consists of devices and means of transmitting bits across computer networks. This includes the specifications for the networking cables and the connectors that join devices together along with specifications describing how signals are sent over these connections.

The most common type of cabling used for connecting computing devices is known as twisted pair. It’s called a twisted pair cable because it features pairs of copper wires that are twisted together.  A standard cat six cable has eight wires consisting of four twisted pairs inside a single jacket.  But in all modern forms of networking, it’s important to know that these cables allow for duplex communication.

Ethernet over twisted pair technologies are the communications protocols that determine how much data can be sent over a twisted pair cable, how quickly that data can be sent, and how long a network cable can be before the data quality begins to degrade. The most common forms of copper twisted-pair cables used in networking, are Cat 5, Cat 5e, and Cat 6 cables. Cat 5e cables have mostly replaced those older Cat 5 cables because their internals reduced crosstalk.

Cat 6 cables can transfer data faster and more reliably than Cat 5e cables can, but because of their internal arrangement, they have a shorter maximum distance when used at higher speeds. The second primary form of networking cable is known as fiber, short for fiber-optic cables.  Fiber cables can transport data faster than copper cables can, but they’re much more expensive and fragile. Fiber can also transport data over much longer distances than copper can without suffering potential data loss, they are also immune to electromagnetic interference.

Twisted pair network cables are terminated with a plug that takes the individual internal wires and exposes them. The most common plug is known as an RJ45, or Registered Jack 45.  A network cable with an RJ45 plug can connect to an RJ45 network port. Network ports are generally directly attached to the devices that make up a computer network. Switches would have many network ports because their purpose is to connect many devices. But servers and desktops, usually only have one or two.

Network devices allow for many computers to communicate with each other. Hubs and switches are the primary devices used to connect computers on a single network, usually referred
to as a LAN, or local area network.

A hub is a physical layer device that allows for connections from many computers at once. All the devices connected to a hub will end up talking to all other devices at the same time. This causes a lot of noise on the network and creates what’s called a collision domain. This causes these systems to have to wait for a quiet period before they try sending their data again. It really slows down network communications and is the primary reason hubs are fairly rare.

In much more common way of connecting many computers is with a more sophisticated device known as a network switch, The difference is that while a hub is a layer one or physical layer device, a switch is a layer two or data link device. This means that a switch can actually inspect the contents of the ethernet protocol data being sent around the network, determine which system
the data is intended for and then only send that data to that one system. This reduces or even completely eliminates the size of collision domains on the network. If you guess that this will lead
to fewer re-transmissions and higher overall throughput,

But we often want to send or receive data to computers on other networks, this is where routers come into play. A router is a device that knows how to forward data between independent networks. While a hub is a layer 1 device and a switch is a layer 2 device, a router operates at layer 3, a network layer. Just like a switch can inspect Ethernet data to determine where to send things, a router can inspect IP data to determine where to send things. Routers store internal tables
containing information about how to route traffic between lots of different networks all over the world.

The most common type of router you’ll see is one for a home network or a small office. These devices generally don’t have very detailed routing tables. The purpose of these routers is mainly
just to take traffic originating from inside the home or office LAN and to forward it along to the ISP, or Internet service provider. Once traffic is at the ISP, a way more sophisticated type of router takes over.

Core ISP routers don’t just handle a lot more traffic than a home or small office router, they also have to deal with much more complexity in making decisions about where to send traffic. A core router usually has many different connections to many other routers. Routers share data with each other via a protocol known as BGP, or border gateway protocol, that let’s them learn about
the most optimal paths to forward traffic.

Datalink Layer

The second layer in our model is known as the data link layer. While the physical layer is all about cabling, connectors and sending signals, the data link layer is responsible for defining a common way of interpreting these signals, so network devices can communicate.

Lots of protocols exist at the data link layer, but the most common is known as Ethernet, although wireless technologies are becoming more and more popular. Beyond specifying physical layer attributes, the Ethernet standards also define a protocol responsible for getting data to nodes on the same network or link.

Ethernet and the data link layer provide a means for software at higher levels of the stack to send and receive data. One of the primary purposes of this layer is to essentially abstract away the need for any other layers to care about the physical layer and what hardware is in use. By dumping this responsibility on the data link layer, the Internet, transport and application layers can all operate the same no matter how the device they’re running on is connected. So, for example, your web browser doesn’t need to know if it’s running on a device connected via a twisted pair or a wireless connection. It just needs the underlying layers to send and receive data for it.

Ethernet, as a protocol, solved the problem of collision by using a technique known as carrier sense multiple access with collision detection abbreviated as CSMA/CD. CSMA/CD is used to determine when the communications channels are clear and when the device is free to transmit data. The way CSMA/CD works is actually pretty simple. If there’s no data currently being transmitted on the network segment, a node will feel free to send data. If it turns out that two or more computers end up trying to send data at the same time, the computers detect this collision and stop sending data. Each device involved with the collision then waits a random interval of time before trying to send data again. This random interval helps to prevent all the computers involved in the collision from colliding again the next time they try to transmit anything.

Ethernet uses MAC addresses to ensure that the data it sends has both an address for the machine that sent the transmission, as well as the one that the transmission was intended for. A MAC address is a globally unique identifier attached to an individual network interface. It’s a 48-bit number normally represented by six groupings of two hexadecimal numbers. Another way to reference each group of numbers in a MAC address is an octet. A MAC address is split into two sections. The first three octets of a MAC address are known as the organizationally unique identifier or OUI. These are assigned to individual hardware manufacturers by the IEEE or the Institute of Electrical and Electronics Engineers.  The last three octets of MAC address can be assigned in any way that the manufacturer would like with the condition that they only assign each possible address once to keep all MAC addresses globally unique.  In this way, even on a network segment, acting as a single collision domain, each node on that network knows when traffic is intended for it.

What is an Ethernet frame? Definition, structure, and variants - IONOS

The first part of an Ethernet frame is known as the preamble. A preamble is 8 bytes or 64 bits long and can itself be split into two sections. The first seven bytes are a series of alternating ones and zeros. These act partially as a buffer between frames and can also be used by the network interfaces to synchronize internal clocks they use, to regulate the speed at which they send data. This last byte in the preamble is known as the SFD or start frame delimiter. This signals to a receiving device that the preamble is over and that the actual frame contents will now follow. Immediately following the start frame delimiter, comes the destination MAC address. This is the hardware address of the intended recipient. Which is then followed by the source MAC address, or where the frame originated from.

The next part of an Ethernet frame is called the EtherType field. It’s 16 bits long and used to describe the protocol of the contents of the frame. Instead of the EtherType field, you could also find what’s known as a VLAN header. It indicates that the frame itself is what’s called a VLAN frame. If a VLAN header is present, the EtherType field follows it. VLAN stands for virtual LAN. VLANs are usually used to segregate different forms of traffic. So you might see a company’s IP
phones operating on one VLAN, while all desktops operate on another.

The data payload of a traditional Ethernet frame can be anywhere from 46 to 1500 bytes long. This contains all of the data from higher layers such as the IP, transport and application layers that’s actually being transmitted. Following that data we have what’s known as a frame check sequence. This is a 4-byte or 32-bit number that represents a checksum value for the entire frame. This checksum value is calculated by performing what’s known as a cyclical redundancy check against the frame.

If the checksum computed by the receiving end doesn’t match the checksum in the frame check sequence field, the data is thrown out. It’s then up to a protocol at a higher layer to decide if that data should be retransmitted. Ethernet itself only reports on data integrity. It doesn’t perform data recovery.


The Network layer: IP

The third layer, the network layer is also sometimes called the Internet layer. While the data link layer is responsible for getting data across a single link, the network layer is responsible for getting data delivered across a collection of networks.

It’s this layer that allows different networks to communicate with each other through devices known as routers. A collection of networks connected together through routers is an internetwork, the most famous of these being the Internet. The most common protocol used at this layer is known as IP or Internet Protocol. IP is the heart of the Internet and most small networks around the world.

The IP network layer is based on a datagram approach, providing only best-effort service, i.e. without any guarantee of the quality of service (QoS). IP is responsible for moving packets of data from router to router according to a four-byte destination IP address (in the IPv4 mode) until the packets reach their destination.

On a local area network or LAN, nodes can communicate with each other through their physical MAC addresses. But MAC addressing isn’t a scheme that scales well.

IP Address: IP addresses are 32-bit long numbers made up of four octets, and each octet is normally described in decimal numbers. IP addresses belong to the networks, not the devices attached to those networks. IP addresses are more hierarchical and easier to store data about than physical addresses are. IP addresses are distributed in large sections to various organizations and companies instead of being determined by hardware vendors. Management and assignment of IP addresses is the responsibility of the Internet authorities.

On many modern networks, you can connect a new device and an IP address will be assigned to it automatically through a technology known as dynamic host configuration protocol. An IP address assigned this way is known as a dynamic IP address. The opposite of this is known as a static IP address, which must be configured on a node manually. In most cases static IP addresses are
reserved for servers and network devices, while dynamic IP addresses are reserved for clients.

Under the IP protocol, a packet is usually referred to as an IP datagram. The two primary sections of an IP datagram are the header and the payload.  The very first field IP datagram header is four bits, and indicates what version of Internet protocol is being used. The most common version of IP is version four or IPv4. Version six or IPv6, is rapidly seeing more widespread adoption.

After the version field, we have the Header Length field. This is also a four bit field that declares how long the entire header is. This is almost always 20 bytes in length when dealing with IPv4. Next, we have the Service Type field. These eight bits can be used to specify details about quality of service or QoS technologies. The important takeaway about QoS is that there are services that allow routers to make decisions about which IP datagram may be more important than others.

The next field is a 16 bit field, known as the Total Length field. It’s used for exactly what it sounds like; to indicate the total length of the IP datagram it’s attached to. The identification field, is a 16-bit number that’s used to group messages together. IP datagrams have a maximum size. Since the Total Length field is 16 bits, the maximum size of a single datagram is the largest number you can represent with 16 bits: 65,535.

If the total amount of data that needs to be sent is larger than what can fit in a single datagram, the IP layer needs to split this data up into many individual packets. When this happens, the identification field is used so that the receiving end understands that every packet with the same value in that field is part of the same transmission.

Next up, we have two closely related fields. The flag field and the Fragmentation Offset field. The flag field is used to indicate if a datagram is allowed to be fragmented, or to indicate that the datagram has already been fragmented. Fragmentation is the process of taking a single IP datagram and splitting it up into several smaller datagrams.

If a datagram has to cross from a network allowing a larger datagram size to one with a smaller datagram size, the datagram would have to be fragmented into smaller ones. The fragmentation offset field contains values used by the receiving end to take all the parts of a fragmented packet and put them back together in the correct order.

The Time to Live or TTL field is an 8-bit field that indicates how many router hops a datagram can traverse before it’s thrown away. Every time a datagram reaches a new router, that router decrements the TTL field by one. Once this value reaches zero, a router knows it doesn’t have to forward the datagram any further. The main purpose of this field is to make sure that when there’s a misconfiguration in routing that causes an endless loop, datagrams don’t spend all eternity trying to reach their destination. An endless loop could be when router A thinks router B is the next hop, and router B thinks router A is the next hop, spoiler alert.

After the TTL field, you’ll find the Protocol field. This is another 8-bit field that contains data about what transport layer protocol is being used. The most common transport layer protocols are TCP and UDP, and we’ll cover both of those in detail in the next few lessons.

So next,  the header checksum field is a checksum of the contents of the entire IP datagram header.  Since the TTL field has to be recomputed at every router that a datagram touches, the checksum field necessarily changes, too.

After all of that, we finally get to two very important fields, the source and destination IP address fields each 32 bits long.  Up next, we have the IP options field. This is an optional field and is used to set special characteristics for datagrams primarily used for testing purposes. The IP options field is usually followed by a padding field. Since the IP options field is both optional and variable in length, the padding field is just a series of zeros used to ensure the header is the correct total size.


The entire contents of an IP datagram are encapsulated as the payload of an Ethernet frame. The contents of IP datagram payload are the entirety of a TCP or UDP packet.


Transport layer: TCP and UDP

While the network layer delivers data between two individual nodes, the transport layer sorts out which client and server programs are supposed to get that data.

The transmission control protocol (TCP) and user datagram protocol (UDP) are transport layer protocols of the Internet protocol reference model. They originate at the end-points of bidirectional communication flows, allowing for end-user terminal services and applications to send and receive data across the Internet.

TCP is responsible for verifying the correct delivery of data between client and server. Data can be lost in the intermediate network. TCP adds support to detect errors or lost data and retransmit them until the data is correctly and completely received. Therefore TCP provides a reliable service through the network underneath may be unreliable, i.e., the operation of Internet protocols does not require reliable transmission of packets, but reliable transmission can reduce the number of retransmissions and thus improve performance.


UDP provides a best-effort service as it does not attempt to recover any error or packet loss. Therefore, it is a protocol providing unreliable transport of user data. But this can be very useful for real-time applications, as re-transmission of any packet may cause more problems than losing packets.


Application layer

The application layer protocols are designed as functions of user terminals or servers. The classic Internet application layer protocols include HTTP for the Web, FTP for file transfer, SMTP for email, Telnet for remote login, DNS for domain name services


Network software is usually divided into client and server categories, with the client application initiating a request for data and the server software answering the request across the network. A single node may be running multiple client or server applications. So, you might run an email program and a web browser, both client applications, on your PC at the same time, and your email and web server might both run on the same server. Even so, emails end up in your email application and web pages end up in your web browser.

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