UDP is aconnectionless protocol, meaning that messages are sent without negotiating a connection and that UDP does not keep track of what it has sent.[1][2] UDP provideschecksums fordata integrity, andport numbers for addressing different functions at the source and destination of the datagram. It has nohandshaking dialogues and thus exposes the user's program to anyunreliability 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 useTransmission Control Protocol (TCP) orStream 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 theprotocol stack. Time-sensitive applications often use UDP because dropping packets is preferable to waiting for packets delayed due toretransmission, which may not be an option in areal-time system.[3]
The protocol was designed byDavid P. Reed in 1980 and formally defined inRFC768.
UDP is a simple message-orientedtransport layer protocol that is documented inRFC 768. Although UDP provides integrity verification (viachecksum) of the header and payload,[4] it provides no guarantees to theupper 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 asUnreliable 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 usedatagram sockets to establish host-to-host communications. An application binds a socket to its endpoint of data transmission, which is a combination of anIP address and aport. In this way, UDP provides applicationmultiplexing. A port is a software structure that is identified by theport 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.
TheInternet Assigned Numbers Authority (IANA) has divided port numbers into three ranges.[6] Port numbers 0 through 1023 are used for common, well-known services. OnUnix-likeoperating systems, using one of these ports requiressuperuser operating permission. Port numbers 1024 through 49151 are theregistered 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 asephemeral ports, which software running on the host may use to dynamically create communications endpoints as needed.[6]
A UDP datagram consists of a datagramheader followed by adata section (the payload data for the application). The UDP datagram header consists of 4 fields, each of which is 2 bytes (16 bits):[3]
The use of theChecksum andSource Port fields is optional in IPv4 (light purple background in table). In IPv6 only theSource Port field is optional. If not used, these fields should be set to zero.[7]
Source Port: 16 bits
This field identifies the sender's port, when used, and should be assumed to be the port to reply to if needed. If the source host is the client, the port number is likely to be an ephemeral port. If the source host is the server, the port number is likely to be awell-known port number from 0 to 1023.[6]
Destination Port: 16 bits
This field identifies the receiver's port and is required. Similar to source port number, if the client is the destination host then the port number will likely be an ephemeral port number and if the destination host is the server then the port number will likely be a well-known port number.[6]
Length: 16 bits
This field specifies the length in bytes of the UDP datagram (the header fields and Data field) inoctets. The minimum length is 8 bytes, the length of the header. The field size sets a theoretical limit of 65,535 bytes (8-byte header + 65,527 bytes of data) for a UDP datagram. However, the actual limit for the data length, which is imposed by the underlyingIPv4 protocol, is 65,507 bytes (65,535 bytes − 8-byte UDP header − 20-byteIP header).[8]
Using IPv6jumbograms it is possible to have UDP datagrams of size greater than 65,535 bytes. The length field is set to zero if the length of the UDP header plus UDP data is greater than 65,535.[9]
The method used to compute the checksum is defined inRFC 768, and efficient calculation is discussed inRFC 1071:
Checksum is the 16-bitones' 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 betweenIPv4 andIPv6 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 apseudo header that contains some of the same information from the realIPv4 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.
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 realIPv6 header:
The checksum is computed over the following fields:
Source address: 128 bits
The address in the IPv6 header.
Destination address: 128 bits
The final destination; if the IPv6 packet does not contain a Routing header, TCP uses the destination address in the IPv6 header, otherwise, at the originating node, it uses the address in the last element of the Routing header, and, at the receiving node, it uses the destination address in the IPv6 header.
UDP length: 32 bits
The length of the UDP header and data (measured in octets).
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 asTFTP, 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 theTransmission 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 andvoice 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.
Voice and video traffic is generally transmitted using UDP. Real-time video andaudio 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 aspoint of sale,accounting, anddatabase systems (when TCP detects packet loss, it will throttle back its data rate usage).[14]
SomeVPN systems such asOpenVPN may use UDP and perform error checking at the application level while implementing reliable connections.WireGuard uses UDP and performs error checking, but does not provide any reliability guarantees, leaving it for the encapsulated protocols to deal with.
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 ofHTTPS which use a combination ofTCP andTLS 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.
Reliable – TCP manages message acknowledgment, retransmission and timeouts. Multiple attempts to deliver the message are made. If data gets lost along the way, data will be re-sent. In TCP, there's either no missing data, or, in case of multiple timeouts, the connection is dropped.
Ordered – If two messages are sent over a connection in sequence, the first message will reach the receiving application first. When data segments arrive in the wrong order, TCP buffers the out-of-order data until all data can be properly re-ordered and delivered to the application.
Heavyweight – TCP requires three packets to set up a socket connection before any user data can be sent. TCP handles reliability andcongestion control.
Streaming – Data is read as abyte stream, no distinguishing indications are transmitted to signal message (segment) boundaries.
User Datagram Protocol is a simpler message-basedconnectionless 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.
Unreliable – When a UDP message is sent, it cannot be known if it will reach its destination; it could get lost along the way. There is no concept of acknowledgment, retransmission, or timeout.
Not ordered – If two messages are sent to the same recipient, the order in which they arrive cannot be guaranteed.
Lightweight – There is no ordering of messages, no tracking connections, etc. It is a very simple transport layer designed on top of IP.
Datagrams – Packets are sent individually and are checked for integrity on arrival. Packets have definite boundaries which are honored upon receipt; a read operation at the receiver socket will yield an entire message as it was originally sent.
No congestion control – UDP itself does not avoid congestion. Congestion control measures must be implemented at the application level or in the network.
Broadcasts – being connectionless, UDP can broadcast - sent packets can be addressed to be receivable by all devices on the subnet.
Multicast – a multicast mode of operation is supported whereby a single datagram packet can be automatically routed without duplication to a group of subscribers.
^Castelli, Matthew J. (2003).Network Sales and Services Handbook. Cisco Press.ISBN9781587050909.
^Stanek, William (2015).Windows Command Line: The Personal Trainer for Windows 8.1 Windows Server 2012 and Windows Server 2012 R2. Stanek & Associates.ISBN9781627164139.
^abcKurose, J. F.; Ross, K. W. (2010).Computer Networking: A Top-Down Approach (5th ed.). Boston, MA: Pearson Education.ISBN978-0-13-136548-3.
^Clark, M.P. (2003).Data Networks IP and the Internet, 1st ed. West Sussex, England: John Wiley & Sons Ltd.
^content@ipv6.com (15 August 2006)."UDP Protocol Overview". Ipv6.com. Retrieved17 August 2011.{{cite web}}: CS1 maint: numeric names: authors list (link)
^abcdeForouzan, B.A. (2000).TCP/IP: Protocol Suite, 1st ed. New Delhi, India: Tata McGraw-Hill Publishing Company Limited.
