Communication protocol | |
Abbreviation | UDP |
---|---|
Developer(s) | David P. Reed |
Introduction | 1980 |
Influenced | QUIC, UDP-Lite |
OSI layer | Transport layer (4) |
RFC(s) | RFC 768 |
Internet protocol suite |
---|
Application layer |
Transport layer |
Internet layer |
Link layer |
In computer networking, the User Datagram Protocol (UDP) is one of the core communication protocols of the Internet protocol suite used to send messages (transported as datagrams in packets) 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.
UDP is a connectionless protocol meaning that messages are sent without negotiating a connection and that UDP doesn't keep track of what it has sent. [1] [2] UDP provides checksums for data integrity, and port numbers for addressing different functions at the source and destination of the datagram. It has no handshaking dialogues and thus exposes the user's program to any unreliability of the underlying network; there is no guarantee of delivery, ordering, or duplicate protection. If error-correction facilities are needed at the network interface level, an application may instead use Transmission Control Protocol (TCP) or Stream Control Transmission Protocol (SCTP) which are designed for this purpose.
UDP is suitable for purposes where error checking and correction are either not necessary or are performed in the application; UDP avoids the overhead of such processing in the protocol stack. Time-sensitive applications often use UDP because dropping packets is preferable to waiting for packets delayed due to retransmission, which may not be an option in a real-time system. [3]
The protocol was designed by David P. Reed in 1980 and formally defined in RFC 768.
UDP is a simple message-oriented transport layer protocol that is documented in RFC 768. Although UDP provides integrity verification (via checksum) of the header and payload, [4] it provides no guarantees to the upper layer protocol for message delivery and the UDP layer retains no state of UDP messages once sent. For this reason, UDP sometimes is referred to as Unreliable Datagram Protocol. [5] If transmission reliability is desired, it must be implemented in the user's application.
A number of UDP's attributes make it especially suited for certain applications.
Applications can use datagram sockets to establish host-to-host communications. An application binds a socket to its endpoint of data transmission, which is a combination of an IP address and a port. In this way, UDP provides application multiplexing. A port is a software structure that is identified by the port number, a 16-bit integer value, allowing for port numbers between 0 and 65535. Port 0 is reserved but is a permissible source port value if the sending process does not expect messages in response.
The Internet Assigned Numbers Authority (IANA) has divided port numbers into three ranges. [6] Port numbers 0 through 1023 are used for common, well-known services. On Unix-like operating systems, using one of these ports requires superuser operating permission. Port numbers 1024 through 49151 are the registered ports used for IANA-registered services. Ports 49152 through 65535 are dynamic ports that are not officially designated for any specific service and may be used for any purpose. These may also be used as ephemeral ports, which software running on the host may use to dynamically create communications endpoints as needed. [6]
A UDP datagram consists of a datagram header followed by a data section (the payload data for the application). The UDP datagram header consists of 4 fields, each of which is 2 bytes (16 bits): [3]
Offset | Octet | 0 | 1 | 2 | 3 | ||||||||||||||||||||||||||||
---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
Octet | Bit | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 | 10 | 11 | 12 | 13 | 14 | 15 | 16 | 17 | 18 | 19 | 20 | 21 | 22 | 23 | 24 | 25 | 26 | 27 | 28 | 29 | 30 | 31 |
0 | 0 | Source Port | Destination Port | ||||||||||||||||||||||||||||||
4 | 32 | Length | Checksum | ||||||||||||||||||||||||||||||
8 | 64 | Data | |||||||||||||||||||||||||||||||
12 | 96 | ||||||||||||||||||||||||||||||||
⋮ | ⋮ |
The use of the Checksum and Source Port fields is optional in IPv4 (light purple background in table). In IPv6 only the Source Port field is optional. If not used, these fields should be set to zero. [7]
The method used to compute the checksum is defined in RFC 768, and efficient calculation is discussed in RFC 1071:
Checksum is the 16-bit ones' complement of the ones' complement sum of a pseudo header of information from the IP header, the UDP header, and the data, padded with zero octets at the end (if necessary) to make a multiple of two octets. [7]
In other words, all 16-bit words are summed using ones' complement arithmetic. Add the 16-bit values up. On each addition, if a carry-out (17th bit) is produced, swing that 17th carry bit around and add it to the least significant bit of the running total. [11] Finally, the sum is then ones' complemented to yield the value of the UDP checksum field.
If the checksum calculation results in the value zero (all 16 bits 0) it should be sent as the ones' complement (all 1s) as a zero-value checksum indicates no checksum has been calculated. [7] In this case, any specific processing is not required at the receiver, because all 0s and all 1s are equal to zero in 1's complement arithmetic.
The differences between IPv4 and IPv6 are in the pseudo header used to compute the checksum, and that the checksum is not optional in IPv6. [12] Under specific conditions, a UDP application using IPv6 is allowed to use a zero UDP zero-checksum mode with a tunnel protocol. [13]
When UDP runs over IPv4, the checksum is computed using a pseudo header that contains some of the same information from the real IPv4 header. [7] : 2 The pseudo header is not the real IPv4 header used to send an IP packet, it is used only for the checksum calculation. UDP checksum computation is optional for IPv4. If a checksum is not used it should be set to the value zero.
Offset | Octet | 0 | 1 | 2 | 3 | ||||||||||||||||||||||||||||
---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
Octet | Bit | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 | 10 | 11 | 12 | 13 | 14 | 15 | 16 | 17 | 18 | 19 | 20 | 21 | 22 | 23 | 24 | 25 | 26 | 27 | 28 | 29 | 30 | 31 |
0 | 0 | Source Address | |||||||||||||||||||||||||||||||
4 | 32 | Destination Address | |||||||||||||||||||||||||||||||
8 | 64 | Zeroes | Protocol | UDP Length | |||||||||||||||||||||||||||||
12 | 96 | Source Port | Destination Port | ||||||||||||||||||||||||||||||
16 | 128 | Length | Checksum | ||||||||||||||||||||||||||||||
20 | 160 | Data | |||||||||||||||||||||||||||||||
24 | 192 | ||||||||||||||||||||||||||||||||
⋮ | ⋮ |
The checksum is calculated over the following fields:
As IPv6 has larger addresses and a different header layout, the method used to compute the checksum is changed accordingly: [10] : §8.1
Any transport or other upper-layer protocol that includes the addresses from the IP header in its checksum computation must be modified for use over IPv6, to include the 128-bit IPv6 addresses instead of 32-bit IPv4 addresses.
When computing the checksum, again a pseudo header is used that mimics the real IPv6 header:
Offset | Octet | 0 | 1 | 2 | 3 | ||||||||||||||||||||||||||||
---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
Octet | Bit | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 | 10 | 11 | 12 | 13 | 14 | 15 | 16 | 17 | 18 | 19 | 20 | 21 | 22 | 23 | 24 | 25 | 26 | 27 | 28 | 29 | 30 | 31 |
0 | 0 | Source address | |||||||||||||||||||||||||||||||
4 | 32 | ||||||||||||||||||||||||||||||||
8 | 64 | ||||||||||||||||||||||||||||||||
12 | 96 | ||||||||||||||||||||||||||||||||
16 | 128 | Destination address | |||||||||||||||||||||||||||||||
20 | 160 | ||||||||||||||||||||||||||||||||
24 | 192 | ||||||||||||||||||||||||||||||||
28 | 224 | ||||||||||||||||||||||||||||||||
32 | 256 | UDP length | |||||||||||||||||||||||||||||||
36 | 288 | Zeroes (0) | Next Header (17) | ||||||||||||||||||||||||||||||
40 | 320 | Source port | Destination port | ||||||||||||||||||||||||||||||
44 | 352 | Length | Checksum | ||||||||||||||||||||||||||||||
48 | 384 | Data | |||||||||||||||||||||||||||||||
52 | 416 | ||||||||||||||||||||||||||||||||
⋮ | ⋮ |
The checksum is computed over the following fields:
Lacking reliability, UDP applications may encounter some packet loss, reordering, errors or duplication. If using UDP, the end-user applications must provide any necessary handshaking such as real-time confirmation that the message has been received. Applications, such as TFTP, may add rudimentary reliability mechanisms into the application layer as needed. [6] If an application requires a high degree of reliability, a protocol such as the Transmission Control Protocol may be used instead.
Most often, UDP applications do not employ reliability mechanisms and may even be hindered by them. Streaming media, real-time multiplayer games and voice over IP (VoIP) are examples of applications that often use UDP. In these particular applications, loss of packets is not usually a fatal problem. In VoIP, for example, latency and jitter are the primary concerns. The use of TCP would cause jitter if any packets were lost as TCP does not provide subsequent data to the application while it is requesting a re-send of the missing data.
Numerous key Internet applications use UDP, including: the Domain Name System (DNS), the Simple Network Management Protocol (SNMP), the Routing Information Protocol (RIP) [3] and the Dynamic Host Configuration Protocol (DHCP).
Voice and video traffic is generally transmitted using UDP. Real-time video and audio streaming protocols are designed to handle occasional lost packets, so only slight degradation in quality occurs, rather than large delays if lost packets were retransmitted. Because both TCP and UDP run over the same network, in the mid-2000s a few businesses found that an increase in UDP traffic from these real-time applications slightly hindered the performance of applications using TCP such as point of sale, accounting, and database systems (when TCP detects packet loss, it will throttle back its data rate usage). [14]
Some VPN systems such as OpenVPN may use UDP and perform error checking at the application level while implementing reliable connections.
QUIC is a transport protocol built on top of UDP. QUIC provides a reliable and secure connection. HTTP/3 uses QUIC as opposed to earlier versions of HTTPS which use a combination of TCP and TLS to ensure reliability and security respectively. This means that HTTP/3 uses a single handshake to set up a connection, rather than having two separate handshakes for TCP and TLS, meaning the overall time to establish a connection is reduced. [15]
Transmission Control Protocol is a connection-oriented protocol and requires handshaking to set up end-to-end communications. Once a connection is set up, user data may be sent bi-directionally over the connection.
User Datagram Protocol is a simpler message-based connectionless protocol. Connectionless protocols do not set up a dedicated end-to-end connection. Communication is achieved by transmitting information in one direction from source to destination without verifying the readiness or state of the receiver.
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.
Internet Protocol version 4 (IPv4) is the first version of the Internet Protocol (IP) as a standalone specification. It is one of the core protocols of standards-based internetworking methods in the Internet and other packet-switched networks. IPv4 was the first version deployed for production on SATNET in 1982 and on the ARPANET in January 1983. It is still used to route most Internet traffic today, even with the ongoing deployment of Internet Protocol version 6 (IPv6), its successor.
Internet Protocol version 6 (IPv6) is the most recent version of the Internet Protocol (IP), the communications protocol that provides an identification and location system for computers on networks and routes traffic across the Internet. IPv6 was developed by the Internet Engineering Task Force (IETF) to deal with the long-anticipated problem of IPv4 address exhaustion, and was intended to replace IPv4. In December 1998, IPv6 became a Draft Standard for the IETF, which subsequently ratified it as an Internet Standard on 14 July 2017.
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.
In computer networking, the maximum transmission unit (MTU) is the size of the largest protocol data unit (PDU) that can be communicated in a single network layer transaction. The MTU relates to, but is not identical to the maximum frame size that can be transported on the data link layer, e.g., Ethernet frame.
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.
Network address translation (NAT) is a method of mapping an IP address space into another by modifying network address information in the IP header of packets while they are in transit across a traffic routing device. The technique was initially used to bypass the need to assign a new address to every host when a network was moved, or when the upstream Internet service provider was replaced but could not route the network's address space. It has become a popular and essential tool in conserving global address space in the face of IPv4 address exhaustion. One Internet-routable IP address of a NAT gateway can be used for an entire private network.
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.
SOCKS is an Internet protocol that exchanges network packets between a client and server through a proxy server. SOCKS5 optionally provides authentication so only authorized users may access a server. Practically, a SOCKS server proxies TCP connections to an arbitrary IP address, and provides a means for UDP packets to be forwarded. A SOCKS server accepts incoming client connection on TCP port 1080, as defined in RFC 1928.
IP fragmentation is an Internet Protocol (IP) process that breaks packets into smaller pieces (fragments), so that the resulting pieces can pass through a link with a smaller maximum transmission unit (MTU) than the original packet size. The fragments are reassembled by the receiving host.
In computer networking, the Datagram Congestion Control Protocol (DCCP) is a message-oriented transport layer protocol. DCCP implements reliable connection setup, teardown, Explicit Congestion Notification (ECN), congestion control, and feature negotiation. The IETF published DCCP as RFC 4340, a proposed standard, in March 2006. RFC 4336 provides an introduction.
Multicast DNS (mDNS) is a computer networking protocol that resolves hostnames to IP addresses within small networks that do not include a local name server. It is a zero-configuration service, using essentially the same programming interfaces, packet formats and operating semantics as unicast Domain Name System (DNS). It was designed to work as either a stand-alone protocol or compatible with standard DNS servers. It uses IP multicast User Datagram Protocol (UDP) packets and is implemented by the Apple Bonjour and open-source Avahi software packages, included in most Linux distributions. Although the Windows 10 implementation was limited to discovering networked printers, subsequent releases resolved hostnames as well. mDNS can work in conjunction with DNS Service Discovery (DNS-SD), a companion zero-configuration networking technique specified separately in RFC 6763.
Robust Header Compression (ROHC) is a standardized method to compress the IP, UDP, UDP-Lite, RTP, and TCP headers of Internet packets.
A network socket is a software structure within a network node of a computer network that serves as an endpoint for sending and receiving data across the network. The structure and properties of a socket are defined by an application programming interface (API) for the networking architecture. Sockets are created only during the lifetime of a process of an application running in the node.
In computer networking, a port or port number is a number assigned to uniquely identify a connection endpoint and to direct data to a specific service. At the software level, within an operating system, a port is a logical construct that identifies a specific process or a type of network service. A port at the software level is identified for each transport protocol and address combination by the port number assigned to it. The most common transport protocols that use port numbers are the Transmission Control Protocol (TCP) and the User Datagram Protocol (UDP); those port numbers are 16-bit unsigned numbers.
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.
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.
The Internet checksum, also called the IPv4 header checksum is a checksum used in version 4 of the Internet Protocol (IPv4) to detect corruption in the header of IPv4 packets. It is carried in the IP packet header, and represents the 16-bit result of summation of the header words.
An IPv6 packet is the smallest message entity exchanged using Internet Protocol version 6 (IPv6). Packets consist of control information for addressing and routing and a payload of user data. The control information in IPv6 packets is subdivided into a mandatory fixed header and optional extension headers. The payload of an IPv6 packet is typically a datagram or segment of the higher-level transport layer protocol, but may be data for an internet layer or link layer instead.
The Stream Control Transmission Protocol (SCTP) is a computer networking communications protocol in the transport layer of the Internet protocol suite. Originally intended for Signaling System 7 (SS7) message transport in telecommunication, the protocol provides the message-oriented feature of the User Datagram Protocol (UDP), while ensuring reliable, in-sequence transport of messages with congestion control like the Transmission Control Protocol (TCP). Unlike UDP and TCP, the protocol supports multihoming and redundant paths to increase resilience and reliability.
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