User Datagram Protocol

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User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. Using UDP, programs on networked computers can send short messages sometimes known as datagrams (using Datagram Sockets) to one another. UDP is sometimes called the Universal Datagram Protocol. The protocol was designed by David P. Reed in 1980.

UDP does not guarantee reliability or ordering in the way that TCP does. Datagrams may arrive out of order, appear duplicated, or go missing without notice. Avoiding the overhead of checking whether every packet actually arrived makes UDP faster and more efficient, for applications that do not need guaranteed delivery. Time-sensitive applications often use UDP because dropped packets are preferable to delayed packets. 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 online games.

[edit] Ports

Main article: List of TCP and UDP port numbers

UDP uses ports to allow application-to-application communication. The port field is a 16 bit value, allowing for port numbers to range 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 (hex 3FF) are named "well-known" ports and on Unix-derived operating systems, binding to one of these ports requires root access.

Ports 1024 through 49,151 (hex BFFF) are registered ports.

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

[edit] Packet structure

UDP is a minimal message-oriented transport layer protocol that is currently documented in IETF RFC 768.

In the Internet protocol suite, UDP provides a very simple interface between a network layer below (e.g., IPv4) and a session layer or application layer above.

UDP provides no guarantees to the upper layer protocol for message delivery and a UDP sender retains no state on UDP messages once sent (for this reason UDP is sometimes called the Unreliable Datagram Protocol). UDP adds only application multiplexing and checksumming of the header and payload. If any kind of reliability for the information transmitted is needed, it must be implemented in upper layers.

The UDP header consists of only 4 fields. The use of two of those is optional (pink background in table).

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 for the data carried by a single 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.

With IPv4

When UDP runs over IPv4, the method used to compute the checksum is defined within 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 together using one's complement (with the checksum field set to zero). The sum is then one's complemented. This final value is then inserted as the checksum field. Algorithmically speaking, this is the same as for IPv6.

The difference is in the data used to make the checksum. Included is 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 IPv4 protocol numbers): 17. The UDP length field is the length of the UDP header and data.

If the checksum is calculated to be zero (all 0s) it should be sent as negative zero (all 1's). If a checksum is not used it should be sent as zero (all 0s) as zero indicates an unused checksum.

With IPv6

When UDP runs over IPv6, the checksum is no longer considered optional, and the method used to compute the checksum is changed, as per 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 instead of 32-bit IPv4 addresses.

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

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 Next Header value is the protocol value for UDP: 17. The UDP length field is the length of the UDP header and data.

If the checksum is calculated to be zero (all 0s) it should be sent as negative zero (all 1's).

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.

While the total amount of UDP traffic found on a typical network is often in the order of only a few percent, numerous key applications use UDP, including: the Domain Name System (DNS), the simple network management protocol (SNMP), the Dynamic Host Configuration Protocol (DHCP) and the Routing Information Protocol (RIP).

[edit] Sample code (Python)

The following, minimalistic example shows how to use UDP for client/server communication:

The server:

import socket
 
PORT = 10000
BUFLEN = 512
 
server = socket.socket(socket.AF_INET, socket.SOCK_DGRAM, socket.IPPROTO_UDP)
server.bind(('', PORT))
 
while True:
        (message, address) = server.recvfrom(BUFLEN)
        print 'Received packet from %s:%d' % (address[0], address[1])
        print 'Data: %s' % message

The client (replace "127.0.0.1" by the IP address of the server):

import socket
 
SERVER_ADDRESS = '127.0.0.1'
SERVER_PORT = 10000
 
client = socket.socket(socket.AF_INET, socket.SOCK_DGRAM, socket.IPPROTO_UDP)
 
for i in range(3):
        print 'Sending packet %d' % i
        message = 'This is packet %d' % i
        client.sendto(message, (SERVER_ADDRESS, SERVER_PORT))
 
client.close()

[edit] Sample code (C++ – Windows-specific)

The following, minimalistic example shows how to use UDP for client/server communication:

The server:

#include <winsock.h>
#include <stdio.h>
#pragma comment(lib,"ws2_32.lib")
 
int main()
{
        WSADATA wsaData;
        SOCKET RecvSocket;
        sockaddr_in RecvAddr;
        int Port = 2345;
        char RecvBuf[1024];
        int  BufLen = 1024;
        sockaddr_in SenderAddr;
        int SenderAddrSize = sizeof(SenderAddr);
        WSAStartup(MAKEWORD(2,2), &wsaData);
        RecvSocket = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
        RecvAddr.sin_family = AF_INET;
        RecvAddr.sin_port = htons(Port);
        RecvAddr.sin_addr.s_addr = INADDR_ANY;
        bind(RecvSocket, (SOCKADDR *) &RecvAddr, sizeof(RecvAddr));
        recvfrom(RecvSocket,RecvBuf, BufLen,0,(SOCKADDR *)&SenderAddr,&SenderAddrSize);
        printf("%s\n",RecvBuf);
        closesocket(RecvSocket);
        WSACleanup();
}

The client:

#include <winsock.h>
#pragma comment(lib,"ws2_32.lib")
 
int main()
{
        WSADATA wsaData;
        SOCKET SendSocket;
        sockaddr_in RecvAddr;
        int Port = 2345;
        char ip[] = "127.0.0.1";
        char SendBuf[] = "hello";
        WSAStartup(MAKEWORD(2,2), &wsaData);
        SendSocket = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
        RecvAddr.sin_family = AF_INET;
        RecvAddr.sin_port = htons(Port);
        RecvAddr.sin_addr.s_addr = inet_addr(ip);
        sendto(SendSocket,SendBuf,strlen(SendBuf)+1,0,(SOCKADDR *) &RecvAddr,sizeof(RecvAddr));
        WSACleanup();
}

[edit] Voice and Video Traffic

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 (if any) occurs rather than large delays as lost packets are 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 bandwidth usage which allows the UDP applications to consume even more bandwidth, worsening the problem. Since both real-time and business applications are both important to businesses, developing quality of service solutions is crucial.^[1]

[edit] Difference between TCP and UDP

TCP ("Transmission Control Protocol") is a connection-oriented protocol, which means that upon communication it requires handshaking to set up end-to-end connection. A connection can be made from client to server, and from then on any data can be sent along that 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 along a connection, one after the other, the first message will reach the receiving application first. When data packets arrive in the wrong order, the TCP layer holds the later data until the earlier data can be rearranged and delivered to the application.
• Heavyweight - TCP requires three packets just to set up a socket, before any actual data can be sent. It handles connections, reliability and congestion control. It is a large transport protocol designed on top of IP.
• Streaming - Data is read as a "stream," with nothing distinguishing where one packet ends and another begins. Packets may be split or merged into bigger or smaller data streams arbitrarily.

UDP is a simpler message-based connectionless protocol. In connectionless protocols, there is no effort made to setup a dedicated end-to-end connection. Communication is achieved by transmitting information in one direction, from source to destination without checking to see if the destination is still there, or if it is prepared to receive the information. With UDP messages (packets) cross the network in independent units.

• 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 guaranteed to be whole if they arrive. Packets have definite bounds and no split or merge into data streams may exist..

[edit] Notes

[edit] See also

[edit] External links

• RFC 768
• IANA Port Assignments
• The Trouble with UDP Scanning (PDF)
• Breakdown of UDP frame
• UDP on MSDN Magazine Sockets and WCF