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.
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.[1] 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).
[2]

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.
Contents [hide]
• 1 Ports
• 2 Packet structure
• 3 Checksum computation
○ 3.1 IPv4 PSEUDO-HEADER
○ 3.2 IPv6 PSEUDO-HEADER
• 4 Reliability and congestion control solutions
• 5 Applications
• 6 Comparison of UDP and TCP
• 7 See also
• 8 References
• 9 External links

Ports
Main article: TCP and UDP port
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.
The Internet Assigned Numbers Authority has divided port numbers into three ranges:[2]
Ports 0 through 1,023 (hexadecimal 0x3FF) are the well-known (permanent) ports, which are assigned and controlled
by IANA. These are usually used as universal port numbers for servers, though there are exceptions.[2] On Unix-like
operating systems, binding to one of these ports requires superuser (root) access.
Ports 1,024 through 49,151 (0xBFFF) are the registered ports. These ports are not assigned or controlled by IANA.
They can only be registered with IANA to prevent duplication.[2]
Ports 49,152 through 65,535 (0xFFFF) are the dynamic (private) ports, which are neither controlled nor registered.
These ports can be used by any process. They are also known as the ephemeral ports, from which the UDP software
running on the client host randomly chooses in order to define itself.[2] 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 is sometimes referred to as Unreliable Datagram Protocol.
[citation needed]

UDP provides application multiplexing (via port numbers) and integrity verification (via checksum) of the header and
payload.[3] If transmission reliability is desired, it must be implemented in the user's application.
bits 0 – 15 16 – 31
0 Source Port Number Destination Port Number
32 Length Checksum

64 Data

The UDP header consists of 4 fields, all of which are 2 bytes (16 bits).[1] 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 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.[2]
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.[2]
Length
A 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
+ 65,527 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 (65,535 − 8 byte UDP header − 20 byte IP header).[2]
Checksum
The checksum field is used for error-checking of the header and data. If the checksum is omitted in IPv4, the
field uses the value all-zeros.[2] This field is not optional for IPv6.[4]
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.[5]
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 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.
The following table defines the PSEUDO-HEADER used only for the checksum calculation.
bits 0–7 8 – 15 16 – 23 24 – 31
0 Source address
32 Destination address
64 Zeros Protocol UDP length
 96 Source Port Destination Port
128 Length Checksum

160 Data

The source and destination addresses are those in the IPv4 header. The protocol is that for UDP (see List of IP
protocol numbers): 17 (0x11). 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.[4]
When computing the checksum, again a PSEUDO-HEADER is used that mimics the real IPv6 header:
bits 0–7 8 – 15 16 – 23 24 – 31
0
32
Source address
64
96
128
160
Destination address
192
224
256 UDP length
288 Zeros Next Header
320 Source Port Destination Port
352 Length Checksum

384 Data

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.[2] 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
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 Routing Information Protocol (RIP)[1] 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, 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.[6]
Comparison of UDP and TCP
Main article: Transport Layer
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 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 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.
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
• UDP Helper Address
References
1. ^ a b c Kurose, J.F. & Ross, K.W. (2010). Computer Networking, 5th ed. Boston, MA: Pearson Education, Inc.
2. ^ a b c d e f g h i j Forouzan, B.A. (2000). TCP/IP: Protocol Suite, 1st ed. New Delhi, India: Tata McGraw-Hill
Publishing Company Limited.
3. ^ Clark, M.P. (2003). Data Networks IP and the Internet, 1st ed. West Sussex, England: John Wiley & Sons
Ltd.
4. ^ a b Deering S. & Hinden R. (December 1998). RFC 2460: Internet Protocol, Version 6 (IPv6) Specification.
Internet Engineering Task Force. Retrieved from http://tools.ietf.org/html/rfc2460
5. ^ Postel, J. (August 1980). RFC 768: User Datagram Protocol. Internet Engineering Task Force. Retrieved
from http://tools.ietf.org/html/rfc768
6. ^ The impact of UDP on Data Applications