The User Datagram Protocol is one of the core members of the Internet protocol suite.
With UDP, computer applications can send messages, in this case referred to as datagrams, to
other hosts on an Internet Protocol network without prior communications to set up special
transmission channels or data paths.
The protocol was designed by David P. Reed in 1980 and formally defined in RFC 768.
UDP uses a simple transmission model with a minimum of protocol mechanism.
It has no handshaking dialogues, and thus exposes any unreliability of the underlying
network protocol to the user's program.
As this is normally IP over unreliable media, there is no guarantee of delivery, ordering,
or duplicate protection.
UDP provides checksums for data integrity, and port numbers for addressing different
functions at the source and destination of the datagram.
UDP is suitable for purposes where error checking and correction is either not necessary or
is 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 or Stream Control Transmission Protocol which
are designed for this purpose.
A number of UDP's attributes make it especially suited for certain applications.
It is transaction-oriented, suitable for simple query-response protocols such as the Domain
Name System or the Network Time Protocol.
It provides datagrams, suitable for modeling other protocols such as in IP tunneling or
Remote Procedure Call and the Network File System.
It is simple, suitable for bootstrapping or other purposes without a full protocol stack,
such as the DHCP and Trivial File Transfer Protocol.
It is stateless, suitable for very large numbers of clients, such as in streaming media applications
for example IPTV The lack of retransmission delays makes it
suitable for real-time applications such as Voice over IP, online games, and many protocols
built on top of the Real Time Streaming Protocol.
Works well in unidirectional communication, suitable for broadcast information such as
in many kinds of service discovery and shared information such as broadcast time or Routing
Information Protocol
Service ports
Applications 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 service port.
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 has divided port numbers into three ranges.
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.
They also are used as ephemeral ports, from which software running on the host may randomly
choose a port in order to define itself.
In effect, they are used as temporary ports primarily by clients when communicating with
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 sometimes is referred to as Unreliable Datagram Protocol.
UDP provides application multiplexing and integrity verification of the header and payload.
If transmission reliability is desired, it must be implemented in the user's application.
The UDP header consists of 4 fields, each of which is 2 bytes.
The use of the fields "Checksum" and "Source port" is optional in IPv4.
In IPv6 only the source port is optional.
Source port number This field identifies the sender's port when
meaningful and should be assumed to be the port to reply to if needed.
If not used, then it should be zero.
If the source host is the client, the port number is likely to be an ephemeral port number.
If the source host is the server, the port number is likely to be a well-known port number.
Destination port number 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.
Length A field that specifies the length in bytes
of the UDP header and UDP 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 for a UDP datagram.
The practical limit for the data length which is imposed by the underlying IPv4 protocol
is 65,507 bytes.
In IPv6 Jumbograms it is possible to have UDP packets of size greater than 65,535 bytes.
RFC 2675 specifies that the length field is set to zero if the length of the UDP header
plus UDP data is greater than 65,535.
Checksum The checksum field is used for error-checking
of the header and data.
If no checksum is generated by the transmitter, 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 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 it should be sent as the one's
complement.
The difference between IPv4 and IPv6 is in the data used to compute the checksum.
IPv4 Pseudo Header 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.
The pseudo header is not the real IPv4 header used to send an IP packet, it is used only
for the checksum calculation.
The source and destination addresses are those in the IPv4 header.
The protocol is that for UDP: 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 Pseudo Header 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, again a pseudo header is used that mimics the real IPv6 header:
The source address is the one in the IPv6 header.
The destination address is the final destination; if the IPv6 packet does not 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 employ reliability mechanisms and may even be hindered
by them.
Streaming media, real-time multiplayer games and voice over IP are examples of applications
that often use UDP.
In these particular applications, loss of packets is not usually a fatal problem.
If an application requires a high degree of reliability, a protocol such as the Transmission
Control Protocol may be used instead.
Applications Numerous key Internet applications use UDP,
including: the Domain Name System, where queries must be fast and only consist of a single
request followed by a single reply packet, the Simple Network Management Protocol, the
Routing Information Protocol and the Dynamic Host Configuration Protocol.
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 seen as crucial by some.
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.
Multiple attempts to 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 delay 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 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.
However, one primary benefit of UDP over TCP is the application to VoIP where latency and
jitter are the primary concerns.
It is assumed in VoIP UDP that the end users provide any necessary real time confirmation
that the message has been received.
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, or 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.
No congestion control – UDP itself does not avoid congestion, unless they implement
congestion control measures at the application level.
See also
Notes and references Notes
RFC references RFC 768 – User Datagram Protocol
RFC 2460 – Internet Protocol, Version 6 Specification
RFC 2675 – IPv6 Jumbograms RFC 4113 – Management Information Base for
the UDP RFC 5405 – Unicast UDP Usage Guidelines
for Application Designers External links
IANA Port Assignments The Trouble with UDP Scanning
Breakdown of UDP frame UDP on MSDN Magazine Sockets and WCF
UDP connections