Network packet

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In telecommunications and computer networking, a network packet is a formatted unit of data carried by a packet-switched network. A packet consists of control information and user data; [1] the latter is also known as the payload . Control information provides data for delivering the payload (e.g., source and destination network addresses, error detection codes, or sequencing information). Typically, control information is found in packet headers and trailers.

Contents

In packet switching, the bandwidth of the transmission medium is shared between multiple communication sessions, in contrast to circuit switching, in which circuits are preallocated for the duration of one session and data is typically transmitted as a continuous bit stream.

Terminology

In the seven-layer OSI model of computer networking, packet strictly refers to a protocol data unit at layer 3, the network layer. [2] A data unit at layer 2, the data link layer, is a frame . In layer 4, the transport layer, the data units are segments and datagrams. Thus, in the example of TCP/IP communication over Ethernet, a TCP segment is carried in one or more IP packets, which are each carried in one or more Ethernet frames.

Architecture

The basis of the packet concept is the postal letter: the header is like the envelope, the payload is the entire content inside the envelope, and the footer would be your signature at the bottom. [3]

Network design can achieve two major results by using packets: error detection and multiple host addressing. [4]

Framing

Communications protocols use various conventions for distinguishing the elements of a packet and for formatting the user data. For example, in Point-to-Point Protocol, the packet is formatted in 8-bit bytes, and special characters are used to delimit elements. Other protocols, like Ethernet, establish the start of the header and data elements by their location relative to the start of the packet. Some protocols format the information at a bit level instead of a byte level. [5]

Contents

A packet may contain any of the following components:

Addresses
The routing of network packets requires two network addresses, the source address of the sending host, and the destination address of the receiving host. [6]
Error detection and correction
Error detection and correction is performed at various layers in the protocol stack. Network packets may contain a checksum, parity bits or cyclic redundancy checks to detect errors that occur during transmission. [6]
At the transmitter, the calculation is performed before the packet is sent. When received at the destination, the checksum is recalculated, and compared with the one in the packet. If discrepancies are found, the packet may be corrected or discarded. Any packet loss due to these discards is dealt with by the network protocol.
In some cases, modifications of the network packet may be necessary while routing, in which cases checksums are recalculated.
Hop limit
Under fault conditions, packets can end up traversing a closed circuit. If nothing was done, eventually the number of packets circulating would build up until the network was congested to the point of failure. Time to live is a field that is decreased by one each time a packet goes through a network hop. If the field reaches zero, routing has failed, and the packet is discarded. [6]
Ethernet packets have no time-to-live field and so are subject to broadcast storms in the presence of a switching loop.
Length
There may be a field to identify the overall packet length. However, in some types of networks, the length is implied by the duration of the transmission. [6]
Protocol identifier
It is often desirable to carry multiple communication protocols on a network. A protocol identifier field specifies a packet's protocol and allows the protocol stack to process many types of packets.
Priority
Some networks implement quality of service which can prioritize some types of packets above others. This field indicates which packet queue should be used; a high-priority queue is emptied more quickly than lower-priority queues at points in the network where congestion is occurring. [6]
Payload
In general, the payload is the data that is carried on behalf of an application. It is usually of variable length, up to a maximum that is set by the network protocol and sometimes the equipment on the route. When necessary, some networks can break a larger packet into smaller packets. [6]

Examples

Internet protocol

IP packets are composed of a header and payload. The header consists of fixed and optional fields. The payload appears immediately after the header. An IP packet has no trailer. However, an IP packet is often carried as the payload inside an Ethernet frame, which has its own header and trailer.

Per the end-to-end principle, IP networks do not provide guarantees of delivery, non-duplication, or in-order delivery of packets. However, it is common practice to layer a reliable transport protocol such as Transmission Control Protocol on top of the packet service to provide such protection.

NASA Deep Space Network

The Consultative Committee for Space Data Systems (CCSDS) packet telemetry standard defines the protocol used for the transmission of spacecraft instrument data over the deep-space channel. Under this standard, an image or other data sent from a spacecraft instrument is transmitted using one or more packets.

MPEG packetized stream

Packetized elementary stream (PES) is a specification associated with the MPEG-2 standard that allows an elementary stream to be divided into packets. The elementary stream is packetized by encapsulating sequential data bytes from the elementary stream between PES packet headers.

A typical method of transmitting elementary stream data from a video or audio encoder is to first create PES packets from the elementary stream data and then to encapsulate these PES packets inside an MPEG transport stream (TS) packets or an MPEG program stream (PS). The TS packets can then be transmitted using broadcasting techniques, such as those used in an ATSC and DVB.

NICAM

In order to provide mono compatibility, the NICAM signal is transmitted on a subcarrier alongside the sound carrier. This means that the FM or AM regular mono sound carrier is left alone for reception by monaural receivers. The NICAM packet (except for the header) is scrambled with a nine-bit pseudo-random bit-generator before transmission. Making the NICAM bitstream look more like white noise is important because this reduces signal patterning on adjacent TV channels.

See also

Related Research Articles

<span class="mw-page-title-main">Asynchronous Transfer Mode</span> Digital telecommunications protocol for voice, video, and data

Asynchronous Transfer Mode (ATM) is a telecommunications standard defined by the American National Standards Institute and International Telecommunication Union Telecommunication Standardization Sector for digital transmission of multiple types of traffic. ATM was developed to meet the needs of the Broadband Integrated Services Digital Network as defined in the late 1980s, and designed to integrate telecommunication networks. It can handle both traditional high-throughput data traffic and real-time, low-latency content such as telephony (voice) and video. ATM provides functionality that uses features of circuit switching and packet switching networks by using asynchronous time-division multiplexing. ATM was seen in the 1990s as a competitor to Ethernet and networks carrying IP traffic as, unlike Ethernet, it was faster and designed with quality-of-service in mind, but it fell out of favor once Ethernet reached speeds of 1 gigabits per second.

The Internet Control Message Protocol (ICMP) is a supporting protocol in the Internet protocol suite. It is used by network devices, including routers, to send error messages and operational information indicating success or failure when communicating with another IP address. For example, an error is indicated when a requested service is not available or that a host or router could not be reached. ICMP differs from transport protocols such as TCP and UDP in that it is not typically used to exchange data between systems, nor is it regularly employed by end-user network applications.

The Internet Protocol (IP) is the network layer communications protocol in the Internet protocol suite for relaying datagrams across network boundaries. Its routing function enables internetworking, and essentially establishes the Internet.

<span class="mw-page-title-main">OSI model</span> Model of communication of seven abstraction layers

The Open Systems Interconnection (OSI) model is a reference model from the International Organization for Standardization (ISO) that "provides a common basis for the coordination of standards development for the purpose of systems interconnection." In the OSI reference model, the communications between systems are split into seven different abstraction layers: Physical, Data Link, Network, Transport, Session, Presentation, and Application.

<span class="mw-page-title-main">Point-to-Point Protocol</span> Data link layer communication protocol

In computer networking, Point-to-Point Protocol (PPP) is a data link layer communication protocol between two routers directly without any host or any other networking in between. It can provide loop detection, authentication, transmission encryption, and data compression.

The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.

The Transmission Control Protocol (TCP) is one of the main protocols of the Internet protocol suite. It originated in the initial network implementation in which it complemented the Internet Protocol (IP). Therefore, the entire suite is commonly referred to as TCP/IP. TCP provides reliable, ordered, and error-checked delivery of a stream of octets (bytes) between applications running on hosts communicating via an IP network. Major internet applications such as the World Wide Web, email, remote administration, and file transfer rely on TCP, which is part of the Transport layer of the TCP/IP suite. SSL/TLS often runs on top of TCP.

In computer networking, the User Datagram Protocol (UDP) is one of the core communication protocols of the Internet protocol suite used to send messages to other hosts on an Internet Protocol (IP) network. Within an IP network, UDP does not require prior communication to set up communication channels or data paths.

A frame is a digital data transmission unit in computer networking and telecommunications. In packet switched systems, a frame is a simple container for a single network packet. In other telecommunications systems, a frame is a repeating structure supporting time-division multiplexing.

<span class="mw-page-title-main">Protocol data unit</span> Unit of information transmitted over a computer network

In telecommunications, a protocol data unit (PDU) is a single unit of information transmitted among peer entities of a computer network. It is composed of protocol-specific control information and user data. In the layered architectures of communication protocol stacks, each layer implements protocols tailored to the specific type or mode of data exchange.

The Address Resolution Protocol (ARP) is a communication protocol used for discovering the link layer address, such as a MAC address, associated with a given internet layer address, typically an IPv4 address. This mapping is a critical function in the Internet protocol suite. ARP was defined in 1982 by RFC 826, which is Internet Standard STD 37.

In the IEEE 802 reference model of computer networking, the logical link control (LLC) data communication protocol layer is the upper sublayer of the data link layer of the seven-layer OSI model. The LLC sublayer acts as an interface between the medium access control (MAC) sublayer and the network layer.

<span class="mw-page-title-main">Transport layer</span> Layer in the OSI and TCP/IP models providing host-to-host communication services for applications

In computer networking, the transport layer is a conceptual division of methods in the layered architecture of protocols in the network stack in the Internet protocol suite and the OSI model. The protocols of this layer provide end-to-end communication services for applications. It provides services such as connection-oriented communication, reliability, flow control, and multiplexing.

The data link layer, or layer 2, is the second layer of the seven-layer OSI model of computer networking. This layer is the protocol layer that transfers data between nodes on a network segment across the physical layer. The data link layer provides the functional and procedural means to transfer data between network entities and may also provide the means to detect and possibly correct errors that can occur in the physical layer.

<span class="mw-page-title-main">Modbus</span> Serial communications protocol

Modbus or MODBUS is a client/server data communications protocol in the application layer. It was originally designed for use with programmable logic controllers (PLCs), but has become a de facto standard communication protocol for communication between industrial electronic devices in a wide range of buses and networks.

In computer networking, jumbo frames are Ethernet frames with more than 1500 bytes of payload, the limit set by the IEEE 802.3 standard. The payload limit for jumbo frames is variable: while 9000 bytes is the most commonly used limit, smaller and larger limits exist. Many Gigabit Ethernet switches and Gigabit Ethernet network interface controllers and some Fast Ethernet switches and Fast Ethernet network interface cards can support jumbo frames.

UDP-Lite is a connectionless protocol that allows a potentially damaged data payload to be delivered to an application rather than being discarded by the receiving station. This is useful as it allows decisions about the integrity of the data to be made in the application layer, where the significance of the bits is understood. UDP-Lite is described in RFC 3828.

In computer networking, an Ethernet frame is a data link layer protocol data unit and uses the underlying Ethernet physical layer transport mechanisms. In other words, a data unit on an Ethernet link transports an Ethernet frame as its payload.

The internet layer is a group of internetworking methods, protocols, and specifications in the Internet protocol suite that are used to transport network packets from the originating host across network boundaries; if necessary, to the destination host specified by an IP address. The internet layer derives its name from its function facilitating internetworking, which is the concept of connecting multiple networks with each other through gateways.

Generic Stream Encapsulation, or GSE for short, is a Data link layer protocol defined by DVB. GSE provides means to carry packet oriented protocols such as IP on top of uni-directional physical layers such as DVB-S2, DVB-T2 and DVB-C2.

References

  1. Stallings, William (2001). "Glossary" . Business Data Communication (4 ed.). Upper Saddle River, New Jersey, USA: Prentice-Hall, Inc. p.  632. ISBN   0-13-088263-1. Packet: A group of bits that includes data plus control information. Generally refers to a network layer (OSI layer 3) protocol data unit.
  2. "OSI Model".
  3. "Understanding The OSI Reference Model: An Analogy", The TCP/IP Guide, archived from the original on 2014-08-09, retrieved 2014-08-09
  4. "Chapter 5 Link Layer". www.msc.uky.edu. Retrieved 2021-10-23.
  5. "Computer Networking : Principles,Protocols and Practice — CNP3www 2014 documentation". www.computer-networking.info. Retrieved 2024-08-05.
  6. 1 2 3 4 5 6 "Network Packet (fundamental unit of information)". NETWORK ENCYCLOPEDIA. 2019-09-22. Network Packet Content. Retrieved 2024-08-05.