It has been suggested that SYN (TCP) be merged into this article or section. (Discuss) Internet protocol suite | Layer | Protocols | | Application | DNS, TLS/SSL, TFTP, FTP, HTTP, IMAP, IRC, NNTP, NTP, POP3, SIP, SMTP, SNMP, SSH, TELNET, BitTorrent, RTP, rlogin, … | | Transport | TCP, UDP, DCCP, SCTP, IL, RUDP, … | | Network | IP (IPv4, IPv6), ICMP, IGMP, ARP, RARP, … | | Data link | Ethernet, Wi-Fi, Token ring, PPP, SLIP, FDDI, ATM, DTM, Frame Relay, SMDS, … | The Transmission Control Protocol (TCP) is a virtual circuit protocol that is one of the core protocols of the Internet protocol suite, often simply referred to as TCP/IP. Using TCP, applications on networked hosts can create connections to one another, over which they can exchange streams of data. The protocol guarantees reliable and in-order delivery of data from sender to receiver. TCP also distinguishes data for multiple connections by concurrent applications (e.g. Web server and e-mail server) running on the same host. Image File history File links Please see the file description page for further information. ...
SYN (synchronize) is a type of packet used by the Transmission Control Protocol (TCP) when initiating a new connection to synchronize the sequence numbers on two connecting computers. ...
The Internet protocol suite is the set of communications protocols that implement the protocol stack on which the Internet and most commercial networks run. ...
The application layer is the 7 th seventh level of the seven-layer OSI model. ...
The domain name system (DNS) stores and associates many types of information with domain names, but most importantly, it translates domain names (computer hostnames) to IP addresses. ...
Secure Sockets Layer (SSL) and its successor, Transport Layer Security (TLS), are cryptographic protocols which provide secure communications on the Internet for such things as web browsing, e-mail, Internet faxing, and other data transfers. ...
Trivial File Transfer Protocol (TFTP) is a very simple file transfer protocol, with the functionality of a very basic form of FTP; it was first defined in 1980. ...
FTP or file transfer protocol is a commonly used protocol for exchanging files over any network that supports the TCP/IP protocol (such as the Internet or an intranet). ...
Hypertext Transfer Protocol (HTTP) is a method used to transfer or convey information on the World Wide Web. ...
The Internet Message Access Protocol (commonly known as IMAP4, and previously called Internet Mail Access Protocol) is an application layer Internet protocol that allows a local client to access e-mail on a remote server. ...
IRC redirects here. ...
The Network News Transfer Protocol or NNTP is an Internet application protocol used primarily for reading and posting Usenet articles, as well as transferring news among news servers. ...
The Network Time Protocol (NTP) is a protocol for synchronising the clocks of computer systems over packet-switched, variable-latency data networks. ...
In computing, local e-mail clients use the Post Office Protocol version 3 (POP3), an application-layer Internet standard protocol, to retrieve e-mail from a remote server over a TCP/IP connection. ...
Session Initiation Protocol (SIP) is a protocol developed by IETF MMUSIC Working Group and proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality. ...
Simple Mail Transfer Protocol (SMTP) is the de facto standard for e-mail transmissions across the Internet. ...
The simple network management protocol (SNMP) forms part of the internet protocol suite as defined by the Internet Engineering Task Force (IETF). ...
To meet Wikipedias quality standards, this article or section may require cleanup. ...
TELNET is a network protocol used on the Internet or local area network LAN connections. ...
The BitTorrent logo BitTorrent is the name of a peer-to-peer (P2P) file distribution protocol, and is the name of a free software implementation of that protocol. ...
The Real-time Transport Protocol (or RTP) defines a good standardized packet format for delivering audio and video over the Internet. ...
In computing, rlogin is a Unix software utility that allows users to log in on another host via a network, communicating via TCP port 513. ...
The User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. ...
The Datagram Congestion Control Protocol (DCCP) is a message-oriented transport layer protocol that is currently under development in the IETF. Applications that might make use of DCCP include those with timingconstraints on the delivery of data such that reliable in-order delivery, when combined with congestion control, is likely...
The Stream Control Transmission Protocol (SCTP) is a transport layer protocol defined in 2000 by the IETF Signaling Transport (SIGTRAN) working group. ...
Transport layer protocol designed originally as part of the Plan 9 from Bell Labs operating system and used to carry 9P. Its main features are: Reliable datagram service In-sequence delivery Internetworking using IP Low complexity, high performance Adaptive timeouts The original paper describing IL: [1] Categories: Computer stubs ...
In computer networking, the Reliable User Datagram Protocol (RUDP) is a transport layer protocol designed at Bell Labs for the Plan 9 operating system. ...
The network layer is level three of the seven level OSI model. ...
The Internet Protocol (IP) is a data-oriented protocol used for communicating data across a packet-switched internetwork. ...
Internet Protocol version 4 is the fourth iteration of the Internet Protocol (IP) and it is the first version of the protocol to be widely deployed. ...
Internet Protocol version 6 (IPv6) is a network layer IP standard used by electronic devices to exchange data across a packet-switched internetwork. ...
The Internet Control Message Protocol (ICMP) is one of the core protocols of the Internet protocol suite. ...
The Internet Group Management Protocol is a communications protocol used to manage the membership of Internet Protocol multicast groups. ...
In computer networking, the Address Resolution Protocol (ARP) is the method for finding a hosts hardware address when only its IP address is known. ...
Reverse address resolution protocol (RARP) is a protocol used to resolve an IP address from a given hardware address (such as an Ethernet address). ...
To meet Wikipedias quality standards, this article or section may require cleanup. ...
Ethernet is a large and diverse family of frame-based computer networking technologies for local area networks (LANs). ...
Wi-Fi (also WiFi, wifi, etc. ...
Token-Ring local area network (LAN) technology was developed and promoted by IBM in the early 1980s and standardised as IEEE 802. ...
In computing, the Point-to-Point Protocol, or PPP, is commonly used to establish a direct connection between two nodes. ...
The Serial Line Internet Protocol (SLIP) is a mostly obsolete encapsulation of the Internet Protocol designed to work over serial ports and modem connections. ...
In computer networking, fiber-distributed data interface (FDDI) is a standard for data transmission in a local area network that can extend in range up to 200 km (124 miles). ...
Asynchronous Transfer Mode (ATM) is a cell relay network protocol which encodes data traffic into small fixed-sized (53 byte; 48 bytes of data and 5 bytes of header information) cells instead of variable sized packets (sometimes known as frames) as in packet-switched networks (such as the Internet Protocol...
Dynamic synchronous Transfer Mode , or DTM for short, is a network protocol. ...
In the context of computer networking, frame relay (also found written as frame-relay) consists of an efficient data transmission technique used to send digital information quickly and cheaply in a relay of frames to one or many destinations from one or many end-points. ...
SMDS, which stands for Switched Multi-megabit Data Services, was a connectionless service used to connect LANs, MANs and WANs to exchange data. ...
A virtual circuit (VC) is a communications arrangement in which data from a source user may be passed to a destination user over more than one real communications circuit during a single period of communication, but the switching is hidden from the users. ...
The Internet protocol suite is the set of communications protocols that implement the protocol stack on which the Internet and most commercial networks run. ...
The Internet protocol suite is the set of communications protocols that implement the protocol stack on which the Internet runs. ...
TCP supports many of the Internet's most popular application protocols and resulting applications, including the World Wide Web, e-mail and Secure Shell. WWWs historical logo designed by Robert Cailliau The World Wide Web (WWW or simply the Web) is a global, read-write information space. ...
Wikipedia does not yet have an article with this exact name. ...
To meet Wikipedias quality standards, this article or section may require cleanup. ...
In the Internet protocol suite, TCP is the intermediate layer between the Internet Protocol (IP) below it, and an application above it. Applications often need reliable pipe-like connections to each other, whereas the Internet Protocol does not provide such streams, but rather only unreliable packets. TCP does the task of the transport layer in the simplified OSI model of computer networks. The other main transport-level Internet protocol is UDP. The Internet protocol suite is the set of communications protocols that implement the protocol stack on which the Internet and most commercial networks run. ...
The Internet Protocol (IP) is a data-oriented protocol used for communicating data across a packet-switched internetwork. ...
Application software is a subclass of computer software that employs the capabilities of a computer directly to a task that the user wishes to perform. ...
In software engineering, a pipeline consisting of chain of processes or other data processing entities, arranged so that the output of each element of the chain is the input of the of the next one. ...
See also: Cox Communications Broadband Internet This definition was formerly listed for connection-oriented, but was refactored due to inaccuracies. ...
A packet is the fundamental unit of information carriage in all modern computer networks that use packet switching. ...
In computing and telecommunications, the transport layer is layer four of the seven layer OSI model. ...
OSI Model The Open Systems Interconnection Reference Model (OSI Reference Model or OSI Model for short) is a layered, abstract description for communications and computer network protocol design, developed as part of the Open Systems Interconnection initiative. ...
A computer network is a system for communication between computers. ...
The User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. ...
Applications send streams of octets (8-bit bytes) to TCP for delivery through the network, and TCP divides the byte stream into appropriately sized segments (usually delineated by the maximum transmission unit (MTU) size of the data link layer of the network the computer is attached to). TCP then passes the resulting packets to the Internet Protocol, for delivery through a network to the TCP module of the entity at the other end. TCP checks to make sure that no packets are lost by giving each packet a sequence number, which is also used to make sure that the data are delivered to the entity at the other end in the correct order. The TCP module at the far end sends back an acknowledgement for packets which have been successfully received; a timer at the sending TCP will cause a timeout if an acknowledgement is not received within a reasonable round-trip time (or RTT), and the (presumably lost) data will then be re-transmitted. The TCP checks that no bytes are damaged by using a checksum; one is computed at the sender for each block of data before it is sent, and checked at the receiver. A bit (binary digit) refers to a digit in the binary numeral system (base 2). ...
A byte is commonly used as a unit of storage measurement in computers, regardless of the type of data being stored. ...
In telecommunications, the term protocol data unit (PDU) has the following meanings: Information that is delivered as a unit among peer entities of a network and that may contain control information, address information, or data. ...
In computer networking, the term Maximum Transmission Unit (MTU) refers to the size (in bytes) of the largest datagram that a given layer of a communications protocol can pass onwards. ...
To meet Wikipedias quality standards, this article or section may require cleanup. ...
In telecommunications, the term round-trip delay time has the following meanings: 1. ...
A checksum is a form of redundancy check, a very simple measure for protecting the integrity of data by detecting errors in data that is sent through space (telecommunications) or time (storage). ...
Protocol operation
An abridged version of the TCP state diagram. Unlike TCP's traditional counterpart — User Datagram Protocol — that can immediately start sending packets, TCP provides connections that need to be established before sending data. TCP connections have three phases: Image File history File links Download high resolution version (1166x792, 12 KB) Summary Licensing File links The following pages on the English Wikipedia link to this file (pages on other projects are not listed): Transmission Control Protocol ...
Image File history File links Download high resolution version (1166x792, 12 KB) Summary Licensing File links The following pages on the English Wikipedia link to this file (pages on other projects are not listed): Transmission Control Protocol ...
The User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. ...
- connection establishment
- data transfer
- connection termination
Before describing these three phases, a note about the various states of a connection end-point or Internet socket: In information processing, a state is the complete set of properties (for example, its energy level, etc. ...
It has been suggested that Ip socket be merged into this article or section. ...
- LISTEN
- SYN-SENT
- SYN-RECEIVED
- ESTABLISHED
- FIN-WAIT-1
- FIN-WAIT-2
- CLOSE-WAIT
- CLOSING
- LAST-ACK
- TIME-WAIT
- CLOSED
- LISTEN
- represents waiting for a connection request from any remote TCP and port. (usually set by TCP servers)
- SYN-SENT
- represents waiting for the remote TCP to send back a TCP packet with the SYN and ACK flags set. (usually set by TCP clients)
- SYN-RECEIVED
- represents waiting for the remote TCP to send back an acknowledgment after have sent back a connection acknowledgment to the remote TCP. (usually set by TCP servers)
- ESTABLISHED
- represents that the port is ready to receive/send data from/to the remote TCP. (set by TCP clients and servers)
- TIME-WAIT
- represents waiting for enough time to pass to be sure the remote TCP received the acknowledgment of its connection termination request. According to RFC 793 a connection can stay in TIME-WAIT for a maximum of four minutes.
Connection establishment To establish a connection, TCP uses a 3-way handshake. Before a client attempts to connect with a server, the server must first bind to a port to open it up for connections: this is called a passive open. Once the passive open is established, a client may initiate an active open. To establish a connection, the 3-way (or 3-step) handshake occurs: In telecommunication and microprocessor systems, the term handshaking has the following meanings: In data communications, a sequence of events governed by hardware or software, requiring mutual agreement of the state of the operational modes prior to information exchange. ...
- The active open is performed by sending a SYN to the server.
- In response, the server replies with a SYN-ACK.
- Finally the client sends an ACK back to the server.
At this point, both the client and server have received an acknowledgement of the connection. SYN (synchronize) is a type of packet used by the Transmission Control Protocol (TCP) when initiating a new connection to synchronize the sequence numbers on two connecting computers. ...
Example: - The initiating host (client) sends a synchronization (SYN flag set) packet to initiate a connection. Any SYN packet holds a Sequence Number. The Sequence Number is a 32-bit field TCP segment header. For example let the Sequence Number value for this session be x.
- The other host receives the packet, records the Sequence Number of x from the client, and replies with an acknowledgment and synchronization (SYN-ACK). The Acknowledgment Number is a 32-bit field in TCP segment header. It contains the next sequence number that this host is expecting to receive (x + 1). The host also initiates a return session. This includes a TCP segment with its own initial Sequence Number value of y.
- The initiating host responds with a next Sequence Number (x+1) and a simple Acknowledgment Number value of y + 1, which is the Sequence Number value of the other host + 1.
Data transfer There are a few key features that set TCP apart from User Datagram Protocol: The User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. ...
- Error-free data transfer
- Ordered-data transfer
- Retransmission of lost packets
- Discarding duplicate packets
- Congestion throttling
In the first two steps of the 3-way handshaking, both computers exchange an initial sequence number (ISN). This number can be arbitrary. This sequence number identifies the order of the bytes sent from each computer so that the data transferred is in order regardless of any fragmentation or disordering that occurs during transmission. For every byte transmitted the sequence number must be incremented. Conceptually, each byte sent is assigned a sequence number and the receiver then sends an acknowledgement back to the sender that effectively states that they received it. What is done in practice is only the first data byte is assigned a sequence number which is inserted in the sequence number field and the receiver sends an acknowledgement value of the next byte they expect to receive. For example, if computer A sends 4 bytes with a sequence number of 100 (conceptually, the four bytes would have a sequence number of 100, 101, 102, & 103 assigned) then the receiver would send back an acknowledgement of 104 since that is the next byte it expects to receive in the next packet. By sending an acknowledgement of 104, the receiver is signaling that it received bytes 100, 101, 102, & 103 correctly. If, by some chance, the last two bytes were corrupted then an acknowledgement value of 102 would be sent since 100 & 101 were received successfully. This would not happen for a packet of 4 bytes but it can happen if, for example, 10,000 bytes are sent in 10 different TCP packets and a packet is lost during transmission. If the first packet is lost then the sender would have to resend all 10,000 bytes since the acknowledgement cannot say that it received bytes 1,000 to 10,000 but only that it expects byte 0 because 0 through 999 were lost. (This issue is addressed in SCTP by adding a selective acknowledgement.) The Stream Control Transmission Protocol (SCTP) is a transport layer protocol defined in 2000 by the IETF Signaling Transport (SIGTRAN) working group. ...
Sequence numbers and acknowledgments cover discarding duplicate packets, retransmission of lost packets, and ordered-data transfer. To assure correctness a checksum field is included (see #Packet structure for details on checksumming). A checksum is a form of redundancy check, a very simple measure for protecting the integrity of data by detecting errors in data that is sent through space (telecommunications) or time (storage). ...
The TCP checksum is a quite weak check by modern standards. Data Link Layers with a high probability of bit error rates may require additional link error correction/detection capabilities. If TCP were to be redesigned today, it would most probably have a 32-bit cyclic redundancy check specified as an error check instead of the current checksum. The weak checksum is partially compensated for by the common use of a CRC or better integrity check at layer 2, below both TCP and IP, such as is used in PPP or the Ethernet frame. However, this does not mean that the 16-bit TCP checksum is redundant: remarkably, surveys of Internet traffic have shown that software and hardware errors that introduce errors in packets between CRC-protected hops are common, and that the end-to-end 16-bit TCP checksum catches most of these simple errors. This is the end-to-end principle at work. A cyclic redundancy check (CRC) is a type of hash function used to produce a checksum â a small, fixed number of bits â against a block of data, such as a packet of network traffic or a block of a computer file. ...
The Open Systems Interconnection Reference Model (OSI Model or OSI Reference Model for short) is a layered abstract description for communications and computer network protocol design, developed as part of the Open Systems Interconnect initiative. ...
In computing, the Point-to-Point Protocol, or PPP, is commonly used to establish a direct connection between two nodes. ...
Ethernet is a large and diverse family of frame-based computer networking technologies for local area networks (LANs). ...
The end-to-end principle is one of the central design principles of the Transmission Control Protocol (TCP) widely used on the Internet. ...
Congestion avoidance The final part to TCP is congestion throttling. Acknowledgements for data sent, or lack of acknowledgements, are used by senders to implicitly interpret network conditions between the TCP sender and receiver. Coupled with timers, TCP senders and receivers can alter the behavior of the flow of data. This is more generally referred to as flow control, congestion control and/or network congestion avoidance. TCP uses a number of mechanisms to achieve high performance and avoid congesting the network (i.e., send data faster than either the network, or the host on the other end, can utilize it). These mechanisms include the use of a sliding window, the slow-start algorithm, the congestion avoidance algorithm, the fast retransmit and fast recovery algorithms, and more. The flow control mechanism is used for controlling the flow of data in a network under well-defined conditions, while congestion control is used for controlling the flow of data when congestion has actually occurred . ...
This article needs to be cleaned up to conform to a higher standard of quality. ...
Network congestion avoidance is a process used in computer networks to avoid congestion. ...
In transmit flow control, sliding window is a variable-duration window that allows a sender to transmit a specified number of data units before an acknowledgement is received or before a specified event occurs. ...
Slow-start is part of the congestion control strategy used by TCP, the data transmission protocol used by many Internet applications, such as HTTP and Secure Shell. ...
Network congestion avoidance is a process used in computer networks to avoid congestion. ...
Fast Retransmit is an enhancement to TCP which reduces the time a sender waits before retransmitting a lost segment. ...
Slow-start is part of the congestion control strategy used by TCP, the data transmission protocol used by many Internet applications, such as HTTP and Secure Shell. ...
Enhancing TCP to reliably handle loss, minimize errors, manage congestion and go fast in very high-speed environments are ongoing areas of research and standards development.
TCP window size
TCP sequence numbers and windows behave very much like a clock. The window, whose width (in bytes) is defined by the receiving host, shifts each time it receives and acks a segment of data. Once it runs out of sequence numbers, it loops back to 0. The TCP receive window size is the amount of received data (in bytes) that can be buffered during a connection. The sending host can send only that amount of data before it must wait for an acknowledgment and window update from the receiving host. Image File history File links Tcp. ...
Image File history File links Tcp. ...
Window scaling For more efficient use of high bandwidth networks, a larger TCP window size may be used. The TCP window size field controls the flow of data and is limited to 2 bytes, or a window size of 65,535 bytes. Since the size field cannot be expanded, a scaling factor is used. The TCP window scale option, as defined in RFC 1323, is an option used to increase the maximum window size from 65,535 bytes to 1 Gigabyte. Scaling up to larger window sizes is a part of what is necessary for TCP Tuning. The TCP window scale option is an option to increase the TCP congestion window size above its maximum value of 65,536 bytes. ...
To meet Wikipedias quality standards, this article or section may require cleanup. ...
The window scale option is used only during the TCP 3-way handshake. The window scale value represents the number of bits to left-shift the 16-bit window size field. The window scale value can be set from 0 (no shift) to 14.
Connection termination The connection termination phase uses, at most, a four-way handshake, with each side of the connection terminating independently. When an endpoint wishes to stop its half of the connection, it transmits a FIN packet, which the other end acknowledges with an ACK. Therefore, a typical teardown requires a pair of FIN and ACK segments from each TCP endpoint. In telecommunication and microprocessor systems, the term handshaking has the following meanings: In data communications, a sequence of events governed by hardware or software, requiring mutual agreement of the state of the operational modes prior to information exchange. ...
A connection can be "half-open", in which case one side has terminated its end, but the other has not. The side that has terminated can no longer send any data into the connection, but the other side can. It is also possible for a 3-way handshake when host A sends a FIN and host B replies with a FIN & ACK (merely combines 2 steps into one) and host A replies with an ACK. This is perhaps the most common method. Finally, it is possible for both hosts to send FINs simultaneously then both just have to ACK. This could possibly be considered a 2-way handshake since the FIN/ACK sequence is done in parallel for both directions.
TCP ports TCP uses the notion of port numbers to identify sending and receiving application end-points on a host, or Internet sockets. Each side of a TCP connection has an associated 16-bit unsigned port number (1-65535) reserved by the sending or receiving application. Arriving TCP data packets are identified as belonging to a specific TCP connection by its sockets, that is, the combination of source host address, source port, destination host address, and destination port. This means that a server computer can provide several clients with several services simultaneously, as long as a client takes care of initiating any simultaneous connections to one destination port from different source ports. It has been suggested that this article or section be merged into Computer port (software). ...
It has been suggested that Ip socket be merged into this article or section. ...
Port numbers are categorized into three basic categories: well-known, registered, and dynamic/private. The well-known ports are assigned by the Internet Assigned Numbers Authority (IANA) and are typically used by system-level or root processes. Well-known applications running as servers and passively listening for connections typically use these ports. Some examples include: FTP (21), TELNET (23), SMTP (25) and HTTP (80). Registered ports are typically used by end user applications as ephemeral source ports when contacting servers, but they can also identify named services that have been registered by a third party. Dynamic/private ports can also be used by end user applications, but are less commonly so. Dynamic/private ports do not contain any meaning outside of any particular TCP connection. For other uses of IANA, see IANA (disambiguation). ...
FTP or file transfer protocol is a commonly used protocol for exchanging files over any network that supports the TCP/IP protocol (such as the Internet or an intranet). ...
TELNET is a network protocol used on the Internet or local area network LAN connections. ...
Simple Mail Transfer Protocol (SMTP) is the de facto standard for email transmission across the Internet. ...
HTTP (for HyperText Transfer Protocol) is the primary method used to convey information on the World Wide Web. ...
Development of TCP TCP is both a complex and evolving protocol. However, while significant enhancements have been made and proposed over the years, its most basic operation has not changed significantly since RFC 793, published in 1981. RFC 1122, Host Requirements for Internet Hosts, clarified a number of TCP protocol implementation requirements. RFC 2581, TCP Congestion Control, one of the most important TCP related RFCs in recent years, describes updated algorithms to be used in order to avoid undue congestion. In 2001, RFC 3168 was written to describe explicit congestion notification (ECN), a congestion avoidance signalling mechanism. In the early 21st century, TCP is typically used in approximately 95% of all Internet packets [citation needed]. Common applications that use TCP include HTTP/HTTPS (World Wide Web), SMTP/POP3/IMAP (e-mail) and FTP (file transfer). Its widespread use is testimony to the original designers that their creation was exceptionally well done. 1981 (MCMLXXXI) was a common year starting on Thursday of the Gregorian calendar. ...
2001: A Space Odyssey. ...
Network congestion avoidance is a process used in computer networks to avoid congestion. ...
Network congestion avoidance is a process used in computer networks to avoid congestion. ...
HTTP (for HyperText Transfer Protocol) is the primary method used to convey information on the World Wide Web. ...
https is a URI scheme which is syntactically identical to the http: scheme normally used for accessing resources using HTTP. Using an https: URL indicates that HTTP is to be used, but with a different default port (443) and an additional encryption/authentication layer between HTTP and TCP. This system...
WWWs historical logo designed by Robert Cailliau The World Wide Web (WWW or simply the Web) is a global, read-write information space. ...
Simple Mail Transfer Protocol (SMTP) is the de facto standard for email transmission across the Internet. ...
Post Office Protocol version 3 (POP3) is an application layer Internet standard protocol used to retrieve email from a remote server to a local client over a TCP/IP connection. ...
The Internet Message Access Protocol (commonly known as IMAP, and previously called Interactive Mail Access Protocol) is an application layer Internet protocol used for accessing email on a remote server from a local client. ...
Wikipedia does not yet have an article with this exact name. ...
FTP or file transfer protocol is a commonly used protocol for exchanging files over any network that supports the TCP/IP protocol (such as the Internet or an intranet). ...
The original TCP congestion control was called TCP Reno, but recently, several alternative congestion control algorithms have been proposed: This article needs to be cleaned up to conform to a higher standard of quality. ...
An extension mechanism TCP Interactive (iTCP) allows applications to subscribe to TCP events and respond accordingly enabling various functional extensions to TCP including application assisted congestion control. HighSpeed TCP also called HSTCP is a new congestion control algorithm for TCP protocol. ...
TCP Vegas is a TCP congestion control, or network congestion avoidance, algorithm that emphasizes packet delay, rather than packet loss, as a signal to help determine the rate at which to send packets. ...
The University of Arizona (UA or U of A) is a land-grant and space-grant public institution of higher education and research located in Tucson, Arizona, United States. ...
TCP Westwood (TCPW), is a sender-side-only modification to TCP NewReno that is intended to better handle large bandwidth-delay product paths (large pipes), with potential packet loss due to transmission or other errors (leaky pipes), and with dynamic load (dynamic pipes). ...
The University of California, Los Angeles, popularly known as UCLA, is a public, coeducational university situated in the neighborhood of Westwood within the city of Los Angeles. ...
BIC TCP (Binary Increase Congestion control) is another implementation of TCP with a optimized congestion control algorithm for high speed networks with high latency (LFN: Long Fat Networks). ...
North Carolina State University is a public, coeducational, extensive research university located in Raleigh, North Carolina, United States. ...
This article or section does not cite its references or sources. ...
This article or section does not cite its references or sources. ...
California Institute of Technology The California Institute of Technology (commonly known as Caltech) is a private, coeducational university located in Pasadena, California, in the United States. ...
The University of Bologna (Italian Alma Mater Studiorum Università di Bologna, UNIBO) is the oldest continually operating degree-granting university in the world, and the second biggest university in Italy. ...
TCP Over Wireless TCP has been optimized for wired networks. Any packet loss is considered to be the result of congestion and the window size is reduced dramatically as a precaution. However, wireless links are known to experience sporadic and usually temporary losses due to fading, shadowing, handoff etc. that cannot be considered congestion. Erroneous back-off of the window size due to wireless packet loss is followed by a congestion avoidance phase with a conservative decrease in window size which causes the radio link to be underutilized. Extensive research has been done on this subject on how to combat these harmful effects. Suggested solutions can be categorized as end-to-end solutions (which require modifications at the client and/or server), link layer solutions (such as RLP in CDMA2000), or proxy based solutions (which require some changes in the network without modifying end nodes). This page meets Wikipedias criteria for speedy deletion. ...
Radio Link Protocol (RLP) is a semi-reliable automatic repeat request (ARQ) protocol used over the air interface. ...
CDMA2000 is a family of third-generation (3G) mobile telecommunications standards that use CDMA, a multiple access scheme for digital radio, to send voice, data, and signalling data (such as a dialed telephone number) between mobile phones and cell sites. ...
Hardware TCP Implementations TCP Offload Engines One way to overcome the processing power requirements of TCP is building hardware implementations of it, widely known as TCP Offload Engines (TOE). The main problem of TOEs is that they are hard to integrate into computing systems, requiring extensive changes in the operating system of the computer or device. The first company to develop such a device was Alacritech. TCP Offload Engine or TOE is a technology for the acceleration of TCP/IP, specifically by moving TCP/IP processing to a separate dedicated sub-system from the main host CPU, the overall system TCP/IP performance is improved. ...
A networking company based in US. Alacritechâs technology is supposed to improve network performance by moving some of a networking workload from a servers general-purpose microprocessor to a specialised chip. ...
Debugging TCP A packet sniffer, which intercepts TCP traffic on a network link, can be useful in debugging networks, network stacks and applications which use TCP by showing the user what packets are passing through a link. Packet sniffers (also known as network or protocol analyzers or Ethernet sniffers) are computer software (usually) or computer hardware that can intercept and log traffic passing over a digital network or part of a network. ...
Alternatives to TCP For many applications TCP is not appropriate. The big problem (at least with normal implementations) is that the application cannot get at the packets coming after a lost packet until the retransmitted copy of the lost packet is received. This causes problems for real-time applications such as streaming multimedia (such as Internet radio), real-time multiplayer games and voice over IP (VoIP) where it is sometimes more useful to get most of the data in a timely fashion than it is to get all of the data in order. Wikipedia does not have an article with this exact name. ...
Web radio (or Internet radio) is a broadcasting service transmitted via the Internet. ...
A typical VoIP Solution A typical analog telephone adapter for connecting an ordinary phone to a VoIP network Voice over Internet Protocol, also called VoIP, IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband is the routing of voice conversations over the Internet or through any other...
Also for embedded systems the complexity of TCP can be a problem. The best known example of this is netbooting which generally uses TFTP. Finally some tricks such as transmitting data between two hosts that are both behind NAT (using STUN or similar systems) are far simpler without a relatively complex protocol like TCP in the way. An embedded system is a special-purpose computer system, which is completely encapsulated by the device it controls. ...
Trivial File Transfer Protocol (TFTP) is a very simple file transfer protocol akin to a basic version of FTP. TFTP is often used to transfer small files between hosts on a network, such as when a remote X Window System terminal or any other thin client boots from a network...
In computer networking, the process of network address translation (NAT, also known as network masquerading or IP-masquerading) involves re-writing the source and/or destination addresses of IP packets as they pass through a router or firewall. ...
STUN is (Simple Traversal Underneath Network Address Translators (NAT)) is a network protocol allowing clients behind NAT (or multiple NATs) to find out its public address, the type of NAT it is behind and the internet side port associated by the NAT with a particular local port. ...
Generally where TCP is unsuitable the User Datagram Protocol (UDP) is used. This provides the application multiplexing and checksums that TCP does, but does not handle building streams or retransmission giving the application developer the ability to code those in a way suitable for the situation and/or to replace them with other methods like forward error correction or interpolation. The User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. ...
In telecommunications, multiplexing (also muxing or MUXing) is the combining of two or more information channels onto a common transmission medium using hardware called a multiplexer or (MUX). ...
It has been suggested that this article or section be merged with Error correction and detection. ...
In the mathematical subfield of numerical analysis, interpolation is a method of constructing new data points from a discrete set of known data points. ...
SCTP is another IP protocol that provides reliable stream oriented services not so dissimilar from TCP. It is newer and considerably more complex than TCP so has not yet seen widespread deployment, however it is especially designed to be used in situations where reliability and near-real-time considerations are important. The Stream Control Transmission Protocol (SCTP) is a transport layer protocol defined in 2000 by the IETF Signaling Transport (SIGTRAN) working group. ...
TCP also has some issues in high bandwidth utilization environments. The TCP congestion avoidance algorithm works very well for ad-hoc environments where it is not known who will be sending data, but if the environment is predictable a timing based protocol such as ATM can avoid the overhead of retransmits that TCP needs. The TCP uses various variations of an additive-increase-multiplicative-decrease (AIMD) scheme, with other schemes such as slow-start in order to achieve congestion avoidance. ...
Asynchronous Transfer Mode (ATM) is a cell relay network protocol which encodes data traffic into small fixed-sized (53 byte; 48 bytes of data and 5 bytes of header information) cells instead of variable sized packets (sometimes known as frames) as in packet-switched networks (such as the Internet Protocol...
Packet structure A TCP packet consists of two sections: The header consists of 11 fields, of which only 10 are required. The eleventh field is optional (pink background in table) and aptly named: options.
Header | + | Bits 0 - 3 | 4 - 9 | 10 - 15 | 16 - 31 | | 0 | Source Port | Destination Port | | 32 | Sequence Number | | 64 | Acknowledgment Number | | 96 | Data Offset | Reserved | Flags | Window | | 128 | Checksum | Urgent Pointer | | 160 | Options (optional) | | 160/192+ | Data | | - Source port
- This field identifies the sending port.
- Destination port
- This field identifies the receiving port.
- Sequence number
- The sequence number has a dual role. If the SYN flag is present then this is the initial sequence number and the first data byte is the sequence number plus 1. Otherwise if the SYN flag is not present then the first data byte is the sequence number.
- Acknowledgement number
- If the ACK flag is set then the value of this field is the sequence number the sender expects next.
- Data offset
- This 4-bit field specifies the size of the TCP header in 32-bit words. The minimum size header is 5 words and the maximum is 15 words thus giving the minimum size of 20 bytes and maximum of 60 bytes. This field gets its name from the fact that it is also the offset from the start of the TCP packet to the data.
- Reserved
- 6-bit reserved field for future use and should be set to zero.
- Flags (aka Control bits)
- This field contains 6 bit flags:
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- URG
- Urgent pointer field is significant
- ACK
- Acknowledgement field is significant
- PSH
- Push function
- RST
- Reset the connection
- SYN
- Synchronize sequence numbers
- FIN
- No more data from sender
- Window
- The number of bytes the sender is willing to receive starting from the acknowledgement field value
- Checksum
- The 16-bit checksum field is used for error-checking of the header and data.
-
- With IPv4
- When TCP runs over IPv4, the method used to compute the checksum is defined in RFC 793:
-
-
- The checksum field is the 16 bit one's complement of the one's complement sum of all 16-bit words in the header and text. If a segment contains an odd number of header and text octets to be checksummed, the last octet is padded on the right with zeros to form a 16-bit word for checksum purposes. The pad is not transmitted as part of the segment. While computing the checksum, the checksum field itself is replaced with zeros.
-
- 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 IPv4.
-
- The difference is in the data used to make the checksum. Included is a pseudo-header that mimics the IPv4 header:
| + | Bits 0 - 3 | 4 - 7 | 8 - 9 | 10 - 15 | 16 - 31 | | 0 | Source address | | 32 | Destination address | | 64 | Zeros | Protocol | TCP length | | 96 | Source Port | Destination Port | | 128 | Sequence Number | | 160 | Acknowledgement Number | | 192 | Data Offset | Reserved | Flags | Window | | 224 | Checksum | Urgent Pointer | | 256 | Options (optional) | | 256/288+ | Data | | -
- The source and destination addresses are those in the IPv4 header. The protocol is that for TCP (see List of IPv4 protocol numbers): 6. The TCP length field is the length of the TCP header and data.
-
- With IPv6
- When TCP runs over IPv6, 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:
| + | Bits 0 - 7 | 8 - 15 | 16 - 23 | 24 - 31 | | 0 | Source address | | 32 | | 64 | | 96 | | 128 | Destination address | | 160 | | 192 | | 256 | | 288 | UDP length | | 320 | Zeros | Next Header | | 352 | Source Port | Destination Port | | 384 | Sequence Number | | 416 | Acknowledgement Number | | 448 | Data Offset | Reserved | Flags | Window | | 480 | Checksum | Urgent Pointer | | 512 | Options (optional) | | 512/544+ | 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 Next Header value is the protocol value for TCP: 6. The TCP length field is the length of the TCP header and data.
- Urgent pointer
- If the URG flag is set, then this 16-bit field is an offset from the sequence number indicating the last urgent data byte.
- Options
- Additional header fields (called options) may follow the urgent pointer. If any options are present then the total length of the option field must be a multiple of a 32-bit word and the data offset field adjusted appropriately.
A checksum is a form of redundancy check, a very simple measure for protecting the integrity of data by detecting errors in data that is sent through space (telecommunications) or time (storage). ...
Internet Protocol version 4 is the fourth iteration of the Internet Protocol (IP) and it is the first version of the protocol to be widely deployed. ...
In mathematics, signed numbers in some arbitrary base is done in the usual way, by prefixing it with a - sign. ...
This is a list of IP protocol numbers that defines the number used in the protocol field of IPv4 packets. ...
Internet Protocol version 6 (IPv6) is a network layer IP standard used by electronic devices to exchange data across a packet-switched internetwork. ...
Data The last field is not a part of the header. The contents of this field are whatever the upper layer protocol wants but this protocol is not set in the header and is presumed based on the port selection.
See also The TCP uses various variations of an additive-increase-multiplicative-decrease (AIMD) scheme, with other schemes such as slow-start in order to achieve congestion avoidance. ...
TCP Vegas is a TCP congestion control, or network congestion avoidance, algorithm that emphasizes packet delay, rather than packet loss, as a signal to help determine the rate at which to send packets. ...
TCP Westwood (TCPW), is a sender-side-only modification to TCP NewReno that is intended to better handle large bandwidth-delay product paths (large pipes), with potential packet loss due to transmission or other errors (leaky pipes), and with dynamic load (dynamic pipes). ...
BIC TCP (Binary Increase Congestion control) is another implementation of TCP with a optimized congestion control algorithm for high speed networks with high latency (LFN: Long Fat Networks). ...
It has been suggested that this article or section be merged into Computer port (software). ...
IANA is responsible for assigning TCP and UDP port numbers to specific uses. ...
In telecommunications, connection-oriented describes a means of transmitting data in which the devices at the end points use a preliminary protocol to establish an end-to-end connection before any data is sent. ...
To meet Wikipedias quality standards, this article or section may require cleanup. ...
T/TCP (Transactional TCP) is a variant of the TCP protocol. ...
Path MTU discovery (PMTUD) is a technique in computing for determining the maximum transmission unit size on the network path between two IP hosts with a view to avoiding IP fragmentation. ...
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