User Datagram Protocol

This article needs additional citations for verification. Please help improve this article by adding reliable references. Unsourced material may be challenged and removed. (April 2009)

The User Datagram Protocol (UDP) is one of the core members of the Internet Protocol Suite, the set of network protocols used for the Internet. With UDP, computer applications can send messages, in this case referred to as datagrams, to other hosts on an Internet Protocol (IP) network without requiring prior communications to set up special transmission channels or data paths. UDP is sometimes called the Universal Datagram Protocol. The protocol was designed by David P. Reed in 1980 and formally defined in RFC 768.^[1]

UDP uses a simple transmission model without implicit hand-shaking dialogues for guaranteeing reliability, ordering, or data integrity. Thus, UDP provides an unreliable service and datagrams may arrive out of order, appear duplicated, or go missing without notice. UDP assumes that error checking and correction is either not necessary or performed in the application, avoiding the overhead of such processing at the network interface level. Time-sensitive applications often use UDP because dropping packets is preferable to waiting for delayed packets, which may not be an option in a real-time system. If error correction facilities are needed at the network interface level, an application may use the Transmission Control Protocol (TCP) or Stream Control Transmission Protocol (SCTP) which are designed for this purpose.

UDP's stateless nature is also useful for servers that answer small queries from huge numbers of clients. Unlike TCP, UDP is compatible with packet broadcast (sending to all on local network) and multicasting (send to all subscribers).

Common network applications that use UDP include: the Domain Name System (DNS), streaming media applications such as IPTV, Voice over IP (VoIP), Trivial File Transfer Protocol (TFTP) and many online games.

Ports

UDP applications use datagram sockets to establish host-to-host communications. Sockets bind the application to service ports, that function as the endpoints of data transmission. 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 65,535. Port 0 is reserved, but is a permissible source port value if the sending process does not expect messages in response.

Ports 1 through 1023 (hexadecimal 0x3FF) are named "well-known" ports and on Unix-like operating systems, binding to one of these ports requires superuser (root) access.

Ports 1024 through 49,151 (0xBFFF) are the registered ports.

Ports 49,152 through 65,535 (0xFFFF) are used as temporary ports primarily by clients when communicating to servers.

Packet structure

UDP is a minimal message-oriented Transport Layer protocol that is documented in IETF RFC 768.

UDP provides no guarantees to the upper layer protocol for message delivery and the UDP protocol layer retains no state of UDP messages once sent. For this reason, UDP is sometimes referred to as Unreliable Datagram Protocol.

UDP provides only application multiplexing (via port numbers) and integrity verification (via checksum) of the header and payload. If any kind of transmission reliability is needed, it must be implemented in the user's application.

The UDP header consists of only 4 fields. The use of two of those is optional in IPv4 (pink background in table). In IPv6 only the source port is optional (see below).

Source port 

This field identifies the sending port when meaningful and should be assumed to be the port to reply to if needed. If not used, then it should be zero.

Destination port 

This field identifies the destination port and is required.

Length 

A 16-bit field that specifies the length in bytes of the entire datagram: header and data. The minimum length is 8 bytes since that's the length of the header. The field size sets a theoretical limit of 65,535 bytes (8 byte header + 65527 bytes of data) for a UDP datagram. The practical limit for the data length which is imposed by the underlying IPv4 protocol is 65,507 bytes.

Checksum 

The 16-bit checksum field is used for error-checking of the header and data. The algorithm for computing the checksum is different for transport over IPv4 and IPv6. If the checksum is omitted in IPv4, the field uses the value all-zeros. This field is not optional for IPv6.

Checksum computation

The method used to compute the checksum is defined in RFC 768:

Checksum is the 16-bit one's complement of the one's 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.

In other words, all 16-bit words are summed using one's complement arithmetic. The sum is then one's 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 one's complement (all 1's).

The difference between IPv4 and IPv6 is in the data used to compute the checksum.

IPv4

When UDP runs over IPv4, the checksum is computed using a pseudo-header that contains information from the IPv4 header:

The source and destination addresses are those in the IPv4 header. The protocol is that for UDP (see List of IP protocol numbers): 17. The UDP length field is the length of the UDP header and data.

UDP checksum computation is optional for IPv4. If a checksum is not used it should be set to the value zero.

IPv6

When UDP runs over IPv6, the checksum is mandatory. The method used to compute it is changed as documented in RFC 2460:

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.

When computing the checksum, a pseudo-header is used that mimics the IPv6 header:

The source address is the one in the IPv6 header. The destination address is the final destination; if the IPv6 packet doesn't contain a Routing header, that will be the destination address in the IPv6 header; otherwise, at the originating node, it will be the address in the last element of the Routing header, and, at the receiving node, it will be the destination address in the IPv6 header. The value of the Next Header field is the protocol value for UDP: 17. The UDP length field is the length of the UDP header and data.

Reliability and congestion control solutions

Lacking reliability, UDP applications must generally be willing to accept some loss, errors or duplication. Some applications such as TFTP may add rudimentary reliability mechanisms into the application layer as needed. Most often, UDP applications do not require 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. If an application requires a high degree of reliability, a protocol such as the Transmission Control Protocol or erasure codes may be used instead.

Lacking any congestion avoidance and control mechanisms, network-based mechanisms are required to minimize potential congestion collapse effects of uncontrolled, high rate UDP traffic loads. In other words, since UDP senders cannot detect congestion, network-based elements such as routers using packet queuing and dropping techniques will often be the only tool available to slow down excessive UDP traffic. The Datagram Congestion Control Protocol (DCCP) is being designed as a partial solution to this potential problem by adding end host TCP-friendly congestion control behavior to high-rate UDP streams such as streaming media.

Applications

While the total amount of UDP traffic found on a typical network is often on the order of only a few percent,^{[citation needed]}, numerous key Internet applications use UDP, including: the Domain Name System (DNS), where queries must be fast and only consist of a single request followed by a single reply packet, the Simple Network Management Protocol (SNMP), the Dynamic Host Configuration Protocol (DHCP) and the Routing Information Protocol (RIP).

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, many businesses are finding that a recent increase in UDP traffic from these real-time applications is hindering 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. Since both real-time and business applications are important to businesses, developing quality of service solutions is crucial.^[2]

Comparison of UDP and TCP

Transmission Control Protocol is a connection-oriented protocol, which means that it 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 timeout. Many attempts to reliably deliver the message are made. If it gets lost along the way, the server will re-request the lost part. 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 and congestion control.
• Streaming – Data is read as a byte stream, no distinguishing indications are transmitted to signal message (segment) boundaries.

UDP 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.

• Unreliable – When a 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 and timeout.
• Not ordered – If two messages are sent to the same recipient, the order in which they arrive cannot be predicted.
• Lightweight – There is no ordering of messages, no tracking connections, etc. It is a small transport layer designed on top of IP.
• Datagrams – Packets are sent individually and are checked for integrity only if they arrive. Packets have definite boundaries which are honored upon receipt, meaning a read operation at the receiver socket will yield an entire message as it was originally sent.

See also

• UDP flood attack
• UDP Data Transport
• UDP Lite, a variant that will deliver packets even if they are malformed
• Reliable User Datagram Protocol (RUDP)
• Transport protocol comparison table

References

External links

• IANA Port Assignments
• The Trouble with UDP Scanning (PDF)
• Breakdown of UDP frame
• UDP on MSDN Magazine Sockets and WCF
• UDP connections