The Address plus Port ( A+P ) within the network layer communications protocol for Internet networking is an experimental approach to the IPv4 address shortage . It is a technique for sharing single IPv4 addresses among several users without using stateful network address translation in the carrier network.
53-553: Normal routing uses the IPv4 address only to identify which host an Internet Protocol packet is destined for. A+P uses the destination Transmission Control Protocol or User Datagram Protocol port in addition to the Internet Protocol address to extend the range of available host addresses. Each host is assigned a unique range of ports which they set as the source port in outbound packets and which they use to receive inbound traffic. A+P
106-435: A retransmission timeout (RTO) that is based on the estimated round-trip time (RTT) between the sender and receiver, as well as the variance in this round-trip time. There are subtleties in the estimation of RTT. For example, senders must be careful when calculating RTT samples for retransmitted packets; typically they use Karn's Algorithm or TCP timestamps. These individual RTT samples are then averaged over time to create
159-470: A FIN packet, which the other end acknowledges with an ACK. Therefore, a typical tear-down requires a pair of FIN and ACK segments from each TCP endpoint. After the side that sent the first FIN has responded with the final ACK, it waits for a timeout before finally closing the connection, during which time the local port is unavailable for new connections; this state lets the TCP client resend the final acknowledgment to
212-514: A PC sends data to a smartphone that is slowly processing received data, the smartphone must be able to regulate the data flow so as not to be overwhelmed. TCP uses a sliding window flow control protocol. In each TCP segment, the receiver specifies in the receive window field the amount of additionally received data (in bytes) that it is willing to buffer for the connection. The sending host can send only up to that amount of data before it must wait for an acknowledgment and receive window update from
265-460: A TCP header creating a TCP segment. The TCP segment is then encapsulated into an Internet Protocol (IP) datagram, and exchanged with peers. The term TCP packet appears in both informal and formal usage, whereas in more precise terminology segment refers to the TCP protocol data unit (PDU), datagram to the IP PDU, and frame to the data link layer PDU: Processes transmit data by calling on
318-429: A TCP segment is retransmitted, it retains the same sequence number as the original delivery attempt. This conflation of delivery and logical data ordering means that, when acknowledgment is received after a retransmission, the sender cannot tell whether the original transmission or the retransmission is being acknowledged, the so-called retransmission ambiguity . TCP incurs complexity due to retransmission ambiguity. If
371-451: A connection before entering the data transfer phase. After data transfer is completed, the connection termination closes the connection and releases all allocated resources. A TCP connection is managed by an operating system through a resource that represents the local end-point for communications, the Internet socket . During the lifetime of a TCP connection, the local end-point undergoes
424-473: A good RTT estimate will be produced—eventually—by waiting until there is an unambiguous acknowledgment before adjusting the RTO. After spurious retransmissions, however, it may take significant time before such an unambiguous acknowledgment arrives, degrading performance in the interim. TCP timestamps also resolve the retransmission ambiguity problem in setting the RTO, though they do not necessarily improve
477-449: A half-duplex close sequence. If the host actively closes a connection, while still having unread incoming data available, the host sends the signal RST (losing any received data) instead of FIN. This assures that a TCP application is aware there was a data loss. A connection can be in a half-open state, in which case one side has terminated the connection, but the other has not. The side that has terminated can no longer send any data into
530-418: A header that includes (among other data) the destination IP address . When the client program on the destination computer receives them, the TCP software in the transport layer re-assembles the segments and ensures they are correctly ordered and error-free as it streams the file contents to the receiving application. Transmission Control Protocol accepts data from a data stream, divides it into chunks, and adds
583-451: A sender to assume an unacknowledged packet to be lost after sufficient time elapses (i.e., determining the RTO time). Retransmission ambiguity can lead a sender's estimate of RTT to be imprecise. In an environment with variable RTTs, spurious timeouts can occur: if the RTT is under-estimated, then the RTO fires and triggers a needless retransmit and slow-start. After a spurious retransmission, when
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#1733105141930636-489: A sequence number by the receiver of data to tell the sender that data has been received to the specified byte. ACKs do not imply that the data has been delivered to the application, they merely signify that it is now the receiver's responsibility to deliver the data. Reliability is achieved by the sender detecting lost data and retransmitting it. TCP uses two primary techniques to identify loss. Retransmission timeout (RTO) and duplicate cumulative acknowledgments (DupAcks). When
689-481: A series of state changes: Before a client attempts to connect with a server, the server must first bind to and listen at a port to open it up for connections: this is called a passive open. Once the passive open is established, a client may establish a connection by initiating an active open using the three-way (or 3-step) handshake: Steps 1 and 2 establish and acknowledge the sequence number for one direction (client to server). Steps 2 and 3 establish and acknowledge
742-407: A single segment (say segment number 100) in a stream is lost, then the receiver cannot acknowledge packets above that segment number (100) because it uses cumulative ACKs. Hence the receiver acknowledges packet 99 again on the receipt of another data packet. This duplicate acknowledgement is used as a signal for packet loss. That is, if the sender receives three duplicate acknowledgments, it retransmits
795-424: A smoothed round trip time (SRTT) using Jacobson's algorithm . This SRTT value is what is used as the round-trip time estimate. 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. As a result, there are a number of TCP congestion avoidance algorithm variations. The maximum segment size (MSS)
848-487: Is connection-oriented , meaning that sender and receiver firstly need to establish a connection based on agreed parameters; they do this through three-way handshake procedure. The server must be listening (passive open) for connection requests from clients before a connection is established. Three-way handshake (active open), retransmission , and error detection adds to reliability but lengthens latency . Applications that do not require reliable data stream service may use
901-477: Is a stateless alternative to conventional stateful network address translation as the A+P gateway does not need to keep track of every Transmission Control Protocol or User Datagram Protocol flow. However, it does require A+P aware software at the end-point, capable of limiting the range of port numbers used for originating connections to its allocated range, either in the end host, or, in the more common likely scenario, in
954-518: Is a sequentially numbered document in a series of technical publications issued by the participants of the early development work groups that created the precursors of the modern Internet . After DARPA began the Internet program in earnest in 1977, the project members were in need of communication and documentation of their work in order to realize the concepts laid out by Bob Kahn and Vint Cerf some years before. The Request for Comments (RFC) series
1007-453: Is the largest amount of data, specified in bytes, that TCP is willing to receive in a single segment. For best performance, the MSS should be set small enough to avoid IP fragmentation , which can lead to packet loss and excessive retransmissions. To accomplish this, typically the MSS is announced by each side using the MSS option when the TCP connection is established. The option value is derived from
1060-557: Is used in PPP or the Ethernet frame. However, introduction of errors in packets between CRC-protected hops is common and the 16-bit TCP checksum catches most of these. TCP uses an end-to-end flow control protocol to avoid having the sender send data too fast for the TCP receiver to receive and process it reliably. Having a mechanism for flow control is essential in an environment where machines of diverse network speeds communicate. For example, if
1113-613: The Internet Protocol Suite . The following Internet Experiment Note (IEN) documents describe the evolution of TCP into the modern version: TCP was standardized in January 1980 as RFC 761 . In 2004, Vint Cerf and Bob Kahn received the Turing Award for their foundational work on TCP/IP. The Transmission Control Protocol provides a communication service at an intermediate level between an application program and
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#17331051419301166-567: The Real-time Transport Protocol (RTP) operating over the User Datagram Protocol (UDP) are usually recommended instead. TCP is a reliable byte stream delivery service that guarantees that all bytes received will be identical and in the same order as those sent. Since packet transfer by many networks is not reliable, TCP achieves this using a technique known as positive acknowledgment with re-transmission . This requires
1219-539: The User Datagram Protocol (UDP) instead, which provides a connectionless datagram service that prioritizes time over reliability. TCP employs network congestion avoidance . However, there are vulnerabilities in TCP, including denial of service , connection hijacking , TCP veto, and reset attack . In May 1974, Vint Cerf and Bob Kahn described an internetworking protocol for sharing resources using packet switching among network nodes. The authors had been working with Gérard Le Lann to incorporate concepts from
1272-546: The User Datagram Protocol : TCP uses a sequence number to identify each byte of data. The sequence number identifies the order of the bytes sent from each computer so that the data can be reconstructed in order, regardless of any out-of-order delivery that may occur. The sequence number of the first byte is chosen by the transmitter for the first packet, which is flagged SYN. This number can be arbitrary, and should, in fact, be unpredictable to defend against TCP sequence prediction attacks . Acknowledgments (ACKs) are sent with
1325-527: The World Wide Web (WWW), email, File Transfer Protocol , Secure Shell , peer-to-peer file sharing , and streaming media . TCP is optimized for accurate delivery rather than timely delivery and can incur relatively long delays (on the order of seconds) while waiting for out-of-order messages or re-transmissions of lost messages. Therefore, it is not particularly suitable for real-time applications such as voice over IP . For such applications, protocols like
1378-546: The customer premises equipment own local NAT44 implementation. This computer networking article is a stub . You can help Misplaced Pages by expanding it . Transmission Control Protocol The Transmission Control Protocol ( TCP ) is one of the main protocols of the Internet protocol suite . It originated in the initial network implementation in which it complemented the Internet Protocol (IP). Therefore,
1431-422: The maximum transmission unit (MTU) size of the data link layer of the networks to which the sender and receiver are directly attached. TCP senders can use path MTU discovery to infer the minimum MTU along the network path between the sender and receiver, and use this to dynamically adjust the MSS to avoid IP fragmentation within the network. Internet Experiment Note An Internet Experiment Note ( IEN )
1484-596: The French CYCLADES project into the new network. The specification of the resulting protocol, RFC 675 ( Specification of Internet Transmission Control Program ), was written by Vint Cerf, Yogen Dalal , and Carl Sunshine, and published in December 1974. It contains the first attested use of the term internet , as a shorthand for internetwork . The Transmission Control Program incorporated both connection-oriented links and datagram services between hosts. In version 4,
1537-441: The Internet Protocol. It provides host-to-host connectivity at the transport layer of the Internet model . An application does not need to know the particular mechanisms for sending data via a link to another host, such as the required IP fragmentation to accommodate the maximum transmission unit of the transmission medium. At the transport layer, TCP handles all handshaking and transmission details and presents an abstraction of
1590-453: The RTT estimate. Sequence numbers allow receivers to discard duplicate packets and properly sequence out-of-order packets. Acknowledgments allow senders to determine when to retransmit lost packets. To assure correctness a checksum field is included; see § Checksum computation for details. The TCP checksum is a weak check by modern standards and is normally paired with a CRC integrity check at layer 2 , below both TCP and IP, such as
1643-415: The TCP and passing buffers of data as arguments. The TCP packages the data from these buffers into segments and calls on the internet module [e.g. IP] to transmit each segment to the destination TCP. A TCP segment consists of a segment header and a data section. The segment header contains 10 mandatory fields, and an optional extension field ( Options , pink background in table). The data section follows
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1696-463: The TCP implementation must perform a lookup on this table to find the destination process. Each entry in the table is known as a Transmission Control Block or TCB. It contains information about the endpoints (IP and port), status of the connection, running data about the packets that are being exchanged and buffers for sending and receiving data. The number of sessions in the server side is limited only by memory and can grow as new connections arrive, but
1749-404: The TCP sender attempts recovery by sending a small packet so that the receiver responds by sending another acknowledgment containing the new window size. If a receiver is processing incoming data in small increments, it may repeatedly advertise a small receive window. This is referred to as the silly window syndrome , since it is inefficient to send only a few bytes of data in a TCP segment, given
1802-519: The acknowledgments for the original transmissions arrive, the sender may believe them to be acknowledging the retransmission and conclude, incorrectly, that segments sent between the original transmission and retransmission have been lost, causing further needless retransmissions to the extent that the link truly becomes congested; selective acknowledgement can reduce this effect. RFC 6298 specifies that implementations must not use retransmitted segments when estimating RTT. Karn's algorithm ensures that
1855-522: The classic RTO discussed below). The time based loss detection algorithm called Recent Acknowledgment (RACK) has been adopted as the default algorithm in Linux and Windows. When a sender transmits a segment, it initializes a timer with a conservative estimate of the arrival time of the acknowledgment. The segment is retransmitted if the timer expires, with a new timeout threshold of twice the previous value, resulting in exponential backoff behavior. Typically,
1908-607: The client must allocate an ephemeral port before sending the first SYN to the server. This port remains allocated during the whole conversation and effectively limits the number of outgoing connections from each of the client's IP addresses. If an application fails to properly close unrequired connections, a client can run out of resources and become unable to establish new TCP connections, even from other applications. Both endpoints must also allocate space for unacknowledged packets and received (but unread) data. The Transmission Control Protocol differs in several key features compared to
1961-407: The connection, but the other side can. The terminating side should continue reading the data until the other side terminates as well. Most implementations allocate an entry in a table that maps a session to a running operating system process. Because TCP packets do not include a session identifier, both endpoints identify the session using the client's address and port. Whenever a packet is received,
2014-546: The entire suite is commonly referred to as TCP/IP . TCP provides reliable , ordered, and error-checked delivery of a stream of octets (bytes) between applications running on hosts communicating via an IP network. Major internet applications such as the World Wide Web , email, remote administration , and file transfer rely on TCP, which is part of the Transport layer of the TCP/IP suite. SSL/TLS often runs on top of TCP. TCP
2067-473: The header and is the payload data carried for the application. The length of the data section is not specified in the segment header; it can be calculated by subtracting the combined length of the segment header and IP header from the total IP datagram length specified in the IP header. TCP protocol operations may be divided into three phases. Connection establishment is a multi-step handshake process that establishes
2120-407: The individual units of data transmission that a message is divided into for efficient routing through the network. For example, when an HTML file is sent from a web server, the TCP software layer of that server divides the file into segments and forwards them individually to the internet layer in the network stack . The internet layer software encapsulates each TCP segment into an IP packet by adding
2173-513: The initial timer value is smoothed RTT + max ( G , 4 × RTT variation ) {\displaystyle {\text{smoothed RTT}}+\max(G,4\times {\text{RTT variation}})} , where G {\displaystyle G} is the clock granularity. This guards against excessive transmission traffic due to faulty or malicious actors, such as man-in-the-middle denial of service attackers . Accurate RTT estimates are important for loss recovery, as it allows
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2226-464: The lack of acknowledgments, are used by senders to infer 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 congestion control or congestion avoidance. Modern implementations of TCP contain four intertwined algorithms: slow start , congestion avoidance , fast retransmit , and fast recovery . In addition, senders employ
2279-417: The last unacknowledged packet. A threshold of three is used because the network may reorder segments causing duplicate acknowledgements. This threshold has been demonstrated to avoid spurious retransmissions due to reordering. Some TCP implementations use selective acknowledgements (SACKs) to provide explicit feedback about the segments that have been received. This greatly improves TCP's ability to retransmit
2332-542: The monolithic Transmission Control Program was divided into a modular architecture consisting of the Transmission Control Protocol and the Internet Protocol . This resulted in a networking model that became known informally as TCP/IP , although formally it was variously referred to as the DoD internet architecture model ( DoD model for short) or DARPA model . Later, it became the part of, and synonymous with,
2385-442: The network connection to the application typically through a network socket interface. At the lower levels of the protocol stack, due to network congestion , traffic load balancing , or unpredictable network behavior, IP packets may be lost , duplicated, or delivered out of order . TCP detects these problems, requests re-transmission of lost data, rearranges out-of-order data and even helps minimize network congestion to reduce
2438-407: The occurrence of the other problems. If the data still remains undelivered, the source is notified of this failure. Once the TCP receiver has reassembled the sequence of octets originally transmitted, it passes them to the receiving application. Thus, TCP abstracts the application's communication from the underlying networking details. TCP is used extensively by many internet applications, including
2491-418: The receiver to respond with an acknowledgment message as it receives the data. The sender keeps a record of each packet it sends and maintains a timer from when the packet was sent. The sender re-transmits a packet if the timer expires before receiving the acknowledgment. The timer is needed in case a packet gets lost or corrupted. While IP handles actual delivery of the data, TCP keeps track of segments –
2544-407: The receiving host. When a receiver advertises a window size of 0, the sender stops sending data and starts its persist timer . The persist timer is used to protect TCP from a deadlock situation that could arise if a subsequent window size update from the receiver is lost, and the sender cannot send more data until receiving a new window size update from the receiver. When the persist timer expires,
2597-504: The relatively large overhead of the TCP header. The final main aspect of TCP is congestion control . TCP uses a number of mechanisms to achieve high performance and avoid congestive collapse , a gridlock situation where network performance is severely degraded. These mechanisms control the rate of data entering the network, keeping the data flow below a rate that would trigger collapse. They also yield an approximately max-min fair allocation between flows. Acknowledgments for data sent, or
2650-625: The right segments. Retransmission ambiguity can cause spurious fast retransmissions and congestion avoidance if there is reordering beyond the duplicate acknowledgment threshold. In the last two decades more packet reordering has been observed over the Internet which led TCP implementations, such as the one in the Linux Kernel to adopt heuristic methods to scale the duplicate acknowledgment threshold. Recently, there have been efforts to completely phase out dupack based fast-retransmissions and replace them with timer based ones. (Not to be confused with
2703-404: The sequence number for the other direction (server to client). Following the completion of these steps, both the client and server have received acknowledgments and a full-duplex communication is established. The connection termination phase uses 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
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#17331051419302756-631: The server in case the ACK is lost in transit. The time duration is implementation-dependent, but some common values are 30 seconds, 1 minute, and 2 minutes. After the timeout, the client enters the CLOSED state and the local port becomes available for new connections. It is also possible to terminate the connection by a 3-way handshake, when host A sends a FIN and host B replies with a FIN & ACK (combining two steps into one) and host A replies with an ACK. Some operating systems, such as Linux and HP-UX , implement
2809-560: Was considered the province of the ARPANET project and the Network Working Group (NWG) which defined the network protocols used on it. Thus, the members of the Internet project decided on publishing their own series of documents, Internet Experiment Notes , which were modeled after the RFCs. Jon Postel became the editor of the new series, in addition to his existing role of administering
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