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Encyclopedia > Transmission Control Protocol
The five-layer TCP/IP model
5. Application layer

DHCP · DNS · FTP · Gopher · HTTP · IMAP4 · IRC · NNTP · XMPP · POP3 · SIP · SMTP · SNMP · SSH · TELNET · RPC · RTCP · RTSP · TLS · SDP · SOAP · GTP · STUN · NTP · (more) The TCP/IP model or Internet reference model, sometimes called the DoD model (DoD, Department of Defense) ARPANET reference model, is a layered abstract description for communications and computer network protocol design. ... The application layer is the seventh level of the seven-layer OSI model. ... DHCP redirects here. ... On the Internet, the Domain Name Server (DNS) associates various sorts of information with so-called domain names; most importantly, it serves as the phone book for the Internet by translating human-readable computer hostnames, e. ... This article is about the File Transfer Protocol standardised by the IETF. For other file transfer protocols, see File transfer protocol (disambiguation). ... Gopher is a distributed document search and retrieval network protocol designed for the Internet. ... Hypertext Transfer Protocol (HTTP) is a communications protocol used to transfer or convey information on intranets and the World Wide Web. ... The Internet Message Access Protocol (commonly known as IMAP or IMAP4, and previously called Internet Mail Access Protocol, Interactive Mail Access Protocol (RFC 1064), and Interim Mail Access Protocol[1]) is an application layer Internet protocol operating on port 143 that allows a local client to access e-mail on... 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. ... Jabber redirects here. ... 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. ... The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. ... 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). ... Secure Shell or SSH is a network protocol that allows data to be exchanged over a secure channel between two computers. ... For the packet switched network, see Telenet. ... Remote procedure call (RPC) is a protocol that allows a computer program running on one computer to cause a subroutine on another computer to be executed without the programmer explicitly coding the details for this interaction. ... RTP Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP). ... The Real Time Streaming Protocol (RTSP), developed by the IETF and created in 1998 as RFC 2326, is a protocol for use in streaming media systems which allows a client to remotely control a streaming media server, issuing VCR-like commands such as play and pause, and allowing time-based... Transport Layer Security (TLS) and its predecessor, Secure Sockets Layer (SSL), are cryptographic protocols that provide secure communications on the Internet for such things as web browsing, e-mail, Internet faxing, instant messaging and other data transfers. ... Session Description Protocol (SDP), is a format for describing streaming media initialization parameters. ... A collection of decorative soaps used for human hygiene purposes. ... GPRS Tunneling Protocol (or GTP) is an IP based protocol used within GSM and UMTS networks. ... STUN (Simple Traversal of UDP over NATs) is a network protocol which helps many types of software and hardware receive UDP data properly through home broadband routers that use network address translation (NAT). ... The Network Time Protocol (NTP) is a protocol for synchronizing the clocks of computer systems over packet-switched, variable-latency data networks. ...

4. Transport layer
TCP · UDP · DCCP · SCTP · RTP · RSVP · IGMP · (more)
3. Network/Internet layer
IP (IPv4 · IPv6) · OSPF · IS-IS · BGP · IPsec · ARP · RARP · RIP · ICMP · ICMPv6 · (more)
2. Data link layer
802.11 · 802.16 · Wi-Fi · WiMAX · ATM · DTM · Token ring · Ethernet · FDDI · Frame Relay · GPRS · EVDO · HSPA · HDLC · PPP · PPTP · L2TP · ISDN · (more)
1. Physical layer
Ethernet physical layer · Modems · PLC · SONET/SDH · G.709 · Optical fiber · Coaxial cable · Twisted pair · (more)
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The Transmission Control Protocol (TCP) is one of the core protocols of the Internet protocol suite. TCP provides reliable, in-order delivery of a stream of bytes, making it suitable for applications like file transfer and e-mail. It is so important in the Internet protocol suite that sometimes the entire suite is referred to as "the TCP/IP protocol suite." In computing and telecommunications, the transport layer is the second highest layer in the four and five layer TCP/IP reference models, where it responds to service requests from the application layer and issues service requests to the Internet layer. ... 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... In the field of computer networking, the IETF Signaling Transport (SIGTRAN) working group defined the Stream Control Transmission Protocol (SCTP) as a transport layer protocol in 2000. ... The Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering audio and video over the Internet. ... The Resource ReSerVation Protocol (RSVP), described in RFC 2205, is a Transport layer protocol designed to reserve resources across a network for an integrated services Internet. ... The Internet Group Management Protocol (IGMP) is a communications protocol used to manage the membership of Internet Protocol multicast groups. ... The network layer is third layer out of seven in OSI model and it is the third layer out of five in TCP/IP 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 protocol for packet-switched internetworks. ... The Open Shortest Path First (OSPF) protocol is a hierarchical interior gateway protocol (IGP) for routing in Internet Protocol, using a link-state in the individual areas that make up the hierarchy. ... Is Is is Yeah Yeah Yeahs third EP, to be released on July 24, 2007. ... The Border Gateway Protocol (BGP) is the core routing protocol of the Internet. ... IPsec (IP security) is a suite of protocols for securing Internet Protocol (IP) communications by authenticating and/or encrypting each IP packet in a data stream. ... In computer networking, the Address Resolution Protocol (ARP) is the standard method for finding a hosts hardware address when only its network layer address is known. ... Reverse Address Resolution Protocol (RARP) is a network layer protocol used to obtain an IP address for a given hardware address (such as an Ethernet address). ... This article is chiefly about the Routing Information Protocol (RIP) for the Internet Protocol, but also discusses some other routing information protocols. ... The Internet Control Message Protocol (ICMP) is one of the core protocols of the Internet protocol suite. ... The ICMP for IPv6 (Internet Control Message Protocol Version 6) is an integral part of the IPv6 architecture and must be completely supported by all IPv6 implementations. ... This article does not cite any references or sources. ... IEEE 802. ... The IEEE 802. ... Official Wi-Fi logo Wi-Fi (pronounced wye-fye, IPA: ) is a wireless technology brand owned by the Wi-Fi Alliance intended to improve the interoperability of wireless local area network products based on the IEEE 802. ... Official WiMax logo WiMAX, the Worldwide Interoperability for Microwave Access, is a telecommunications technology aimed at providing wireless data over long distances in a variety of ways, from point-to-point links to full mobile cellular type access. ... Asynchronous Transfer Mode (ATM) is a cell relay, packet switching network and data link layer protocol which encodes data traffic into small (53 bytes; 48 bytes of data and 5 bytes of header information) fixed-sized cells. ... Dynamic synchronous Transfer Mode , or DTM for short, is a network protocol. ... Token-Ring local area network (LAN) technology was developed and promoted by IBM in the early 1980s and standardised as IEEE 802. ... Ethernet is a large, diverse family of frame-based computer networking technologies that operate at many speeds for local area networks (LANs). ... 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). ... In the context of computer networking, 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. ... General Packet Radio Service (GPRS) is a Mobile Data Service available to users of Global System for Mobile Communications (GSM) and IS-136 mobile phones. ... Evolution-Data Optimized or Evolution-Data only, abbreviated as EV-DO or EVDO and often EV, is one telecommunications standard for the wireless transmission of data through radio signals, typically for broadband Internet access. ... High-Speed Packet Access (HSPA) is a collection of mobile telephony protocols that extend and improve the performance of existing UMTS protocols. ... High-Level Data Link Control (HDLC) is a bit-oriented synchronous data link layer protocol developed by the International Organization for Standardization (ISO). ... In computing, the Point-to-Point Protocol, or PPP, is commonly used to establish a direct connection between two nodes. ... The Point-to-Point Tunneling Protocol (PPTP) is a method for implementing virtual private networks. ... In computer networking, the Layer 2 Tunneling Protocol (L2TP) is a tunneling protocol used to support virtual private networks (VPNs). ... ISDN redirects here. ... This article does not cite any references or sources. ... IEEE photograph of a diagram with the original terms for describing Ethernet drawn by Robert M. Metcalfe around 1976. ... For other uses, see Modem (disambiguation). ... For other uses, see Power band. ... It has been suggested that this article be split into articles entitled Synchronous optical networking, SONET and Synchronous digital hierarchy. ... There are very few or no other articles that link to this one. ... Optical fibers An optical fiber (or fibre) is a glass or plastic fiber designed to guide light along its length. ... Coaxial Cable For the weapon, see coaxial weapon. ... 25 Pair Color Code Chart 10BASE-T UTP Cable Twisted pair cabling is a common form of wiring in which two conductors are wound around each other for the purposes of cancelling out electromagnetic interference known as crosstalk. ... The Internet protocol suite is the set of communications protocols that implement the protocol stack on which the Internet and most commercial networks run. ... 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). ... Wikipedia does not yet have an article with this exact name. ...

Contents

Reason for TCP

The Internet Protocol (IP) works by exchanging groups of information called packets. Packets are short sequences of bytes consisting of a header and a body. The header describes the packet's destination, which routers on the Internet use to pass the packet along, in generally the right direction, until it arrives at its final destination. The body contains the application data. In information technology, a packet is a formatted block of data carried by a packet mode computer network. ... This article refers to the unit of binary information. ... This article describes the computer networking device. ...


In cases of congestion, the IP can discard packets, and, for efficiency reasons, two consecutive packets on the internet can take different routes to the destination. Then, the packets can arrive at the destination in the wrong order.


The TCP software libraries use the IP and provides a simpler interface to applications by hiding most of the underlying packet structures, rearranging out-of-order packets, minimizing network congestion, and re-transmitting discarded packets. Thus, TCP very significantly simplifies the task of writing network applications.


Applicability of TCP

TCP is used extensively by many of the Internet's most popular application protocols and resulting applications, including the World Wide Web, E-mail, File Transfer Protocol, Secure Shell, and some streaming media applications. WWWs historical logo designed by Robert Cailliau The World Wide Web (commonly shortened to the Web) is a system of interlinked, hypertext documents accessed via the Internet. ... Wikipedia does not yet have an article with this exact name. ... This article is about the File Transfer Protocol standardised by the IETF. For other file transfer protocols, see File transfer protocol (disambiguation). ... Secure Shell or SSH is a network protocol that allows data to be exchanged over a secure channel between two computers. ... Streaming media is multimedia that is continuously received by, and normally displayed to, the end-user while it is being delivered by the provider. ...


However, because TCP is optimized for accurate delivery rather than timely delivery, TCP sometimes incurs long delays while waiting for out-of-order messages or retransmissions of lost messages, and it is not particularly suitable for real-time applications such as Voice over IP. For such applications, protocols like the Real-time Transport Protocol (RTP) running over the User Datagram Protocol (UDP) are usually recommended instead[1]. An overview of how VoIP works A typical analog telephone adapter for connecting an ordinary phone to a VoIP network Ciscos implementation of VoIP - IP Phone Voice over Internet Protocol, also called VoIP (pronounced voyp), IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband is the... The Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering audio and video over the Internet. ... User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. ...


Using TCP

Using TCP, applications on networked hosts can create connections to one another, over which they can exchange streams of data using Stream Sockets. TCP also distinguishes data for multiple connections by concurrent applications (e.g., Web server and e-mail server) running on the same host. Stream socket is a type of internet socket which provides a connection-oriented, sequenced, and unduplicated flow of data without record boundaries, with well-defined mechanisms for creating and destroying connections and for detecting errors. ...


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 best effort delivery (i.e., unreliable packets). TCP does the task of the transport layer in the simplified OSI model of computer networks. The other main transport-level Internet protocols are UDP and SCTP. 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. ... Best effort delivery describes a network service in which the network does not provide any guarantees that data is delivered or that a user is given a guaranteed quality of service level or a certain priority. ... In computer networking, a reliable protocol is one that ensures data arrival via some internal method, as opposed to an unreliable protocol, which does not guarantee that all the data will arrive intact (or indeed, at all). ... In information technology, a packet is a formatted block of data carried by a packet mode computer network. ... In computing and telecommunications, the transport layer is the second highest layer in the four and five layer TCP/IP reference models, where it responds to service requests from the application layer and issues service requests to the Internet layer. ... The Open Systems Interconnection Basic 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 (OSI) initiative. ... A computer network is an interconnection of a group of computers. ... User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. ... In the field of computer networking, the IETF Signaling Transport (SIGTRAN) working group defined the Stream Control Transmission Protocol (SCTP) as a transport layer protocol in 2000. ...


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 to which the computer is attached). 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 is delivered to the entity at the other end in the correct order. The TCP module at the far end sends back an acknowledgment for packets which have been successfully received; a timer at the sending TCP will cause a timeout if an acknowledgment 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 corrupted by using a checksum; one is computed at the sender for each block of data before it is sent, and checked at the receiver. This article is about the unit of information. ... For the computer industry magazine, see Byte (magazine). ... Transmission Control Protocol (TCP) accepts data from a data stream, segments it into chunks, and adds a TCP header creating a The tcp segment is then encapsulated, into an IP datagram. ... 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. ... This article does not cite any references or sources. ... In telecommunications, the term round-trip delay time has the following meanings: 1. ... A checksum is a form of redundancy check, a simple way to protect the integrity of data by detecting errors in data that are sent through space (telecommunications) or time (storage). ...


TCP segment structure

A TCP segment consists of two sections:

  • header
  • data

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.

TCP Header
Bit offset Bits 0–3 4–7 8–15 16–31
0 Source port Destination port
32 Sequence number
64 Acknowledgment number
96 Data offset Reserved CWR ECE URG ACK PSH RST SYN FIN Window
128 Checksum Urgent pointer
160 Options (optional)
160/192+  
Data
 
  • Source port(16 bits) – identifies the sending port
  • Destination port(16 bits) – identifies the receiving port
  • Sequence number(32 bits) – 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
  • if the SYN flag is not present then the first data byte is the sequence number
  • Acknowledgment number(32 bits) – if the ACK flag is set then the value of this field is the sequence number that the sender of the acknowledgment expects next.
  • Data offset(4 bits) – 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(4 bits) – for future use and should be set to zero
  • Flags(8 bits) (aka Control bits) – contains 8 bit flags
  • CWR(1 bit) – Congestion Window Reduced (CWR) flag is set by the sending host to indicate that it received a TCP segment with the ECE flag set (added to header by RFC 3168).
  • ECE (ECN-Echo)(1 bit) – indicate that the TCP peer is ECN capable during 3-way handshake (added to header by RFC 3168).
  • URG(1 bit) – indicates that the URGent pointer field is significant
  • ACK(1 bit) – indicates that the ACKnowledgment field is significant
  • PSH(1 bit) – Push function
  • RST(1 bit) – Reset the connection
  • SYN(1 bit) – Synchronize sequence numbers
  • FIN(1 bit) – No more data from sender
  • Window(16 bits) – the number of bytes that may be received on the receiving side before being halted from sliding any further and receiving any more bytes as a result of a packet at the beginning of the sliding window not having been acknowledged or received. Starts at acknowledgement field.
  • Checksum(16 bits) – The 16-bit checksum field is used for error-checking of the header and data
  • Urgent pointer(16 bits) – 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(Variable bits) – the total length of the option field must be a multiple of a 32-bit word and the data offset field adjusted appropriately
  • 0 - End of options list
  • 1 - No operation (NOP, Padding)
  • 2 - Maximum segment size
  • 3 - Window scale
  • 4 - Selective Acknowlegement ok
  • 5 -
  • 6 -
  • 7 -
  • 8 - Timestamp [2]

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. Network congestion avoidance is a process used in computer networks to avoid congestion. ... A checksum is a form of redundancy check, a simple way to protect the integrity of data by detecting errors in data that are sent through space (telecommunications) or time (storage). ...

  • Data (Variable bits): As you might expect, this is the payload, or data portion of an TCP packet. The payload may be any number of application layer protocols. The most common are HTTP, Telnet, SSH, FTP, but other popular protocols also use TCP.

Protocol operation

Unlike TCP's traditional counterpart, User Datagram Protocol, which can immediately start sending packets, TCP provides connections that need to be established before sending data. TCP connections have three phases: User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. ...

  1. connection establishment
  2. data transfer
  3. 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. ... An Internet socket (or commonly, a socket or network socket), is a communication end-point unique to a machine communicating on an Internet Protocol-based network, such as the Internet. ...

  1. LISTEN
  2. SYN-SENT
  3. SYN-RECEIVED
  4. ESTABLISHED
  5. FIN-WAIT-1
  6. FIN-WAIT-2
  7. CLOSE-WAIT
  8. CLOSING
  9. LAST-ACK
  10. TIME-WAIT
  11. 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 having 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 three-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 three-way (or 3-step) handshake occurs: In information technology, telecommunications, and related fields, handshaking is an automated process of negotiation that dynamically sets parameters of a communications channel established between two entities before normal communication over the channel begins. ...

  1. The active open is performed by the client sending a SYN to the server.
  2. In response, the server replies with a SYN-ACK.
  3. Finally the client sends an ACK back to the server.

At this point, both the client and server have received an acknowledgment of the connection.


Example:

  1. The initiating host (client) sends a synchronization packet (SYN flag set to 1) to initiate a connection. It sets the packet's sequence number to a random value x.
  2. The other host receives the packet, records the sequence number x from the client, and replies with an acknowledgment and synchronization (SYN-ACK). The Acknowledgment 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 of value y.
  3. The initiating host responds with the 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.

Vulnerabilities

Vulnerability to Denial of Service

By using a spoofed IP address and repeatedly sending SYN packets attackers can cause the server to consume large amounts of resources keeping track of the bogus connections. This is known as a SYN flood attack. Proposed solutions to this problem include SYN cookies and Cryptographic puzzles. A normal connection between a user (Alice) and a server. ... SYN Cookies are the key element of a technique used to guard against SYN flood attacks. ...


Connection hijacking

An attacker who is able to eavesdrop a TCP session and redirect packets can hijack a TCP connection. To do so, the attacker learns the sequence number from the ongoing communication and forges a false packet that looks like the next packet in the stream. Such a simple hijack can result in one packet being erroneously accepted at one end. When the receiving host acknowledges the extra packet to the other side of the connection, synchronization is lost. Hijacking might be combined with ARP or routing attacks that allow taking control of the packet flow, so as to get permanent control of the hijacked TCP connection.[2]


Impersonating a different IP address was possible prior to RFC 1948, when the initial sequence number was easily guessable. That allowed an attacker to blindly send a sequence of packets that the receiver would believe to come from a different IP, without the need to deploy ARP or routing attacks: it is enough to ensure that the legitimate host of the impersonated IP is down, or bring it to that condition using denial of service attacks.


Data transfer

There are a few key features that set TCP apart from User Datagram Protocol: User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. ...

  • Ordered data transfer
  • Retransmission of lost packets
  • Discarding duplicate packets
  • Error-free data transfer
  • Congestion/Flow control

Ordered data transfer, retransmission of lost packets and discarding duplicate packets

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 acknowledgment 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 acknowledgment 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 acknowledgment of 104 since that is the next byte it expects to receive in the next packet. By sending an acknowledgment 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 acknowledgment value of 102 would be sent since 100 & 101 were received successfully.


However, a problem can occasionally arise when packets are lost. For example, 10,000 bytes are sent in 10 different TCP packets, and the first packet is lost during transmission. The sender would then have to resend all 10,000 bytes; the recipient cannot say that it received bytes 1,000 to 9,999 but only that it failed to receive the first packet, containing bytes 0 to 999. In order to solve this problem, an option of selective acknowledgment (SACK) has been added. This option allows the receiver to acknowledge isolated blocks of packets that were received correctly, rather than the sequence number of the last packet received successively, as in the basic TCP acknowledgment. Each block is conveyed by the starting and ending sequence numbers. In the example above, the receiver would send SACK with sequence numbers 1,000 and 10,000. The sender will thus retransmit only the first packet.


The SACK option is not mandatory and it is used only if both parties support it. This is negotiated when connection is established. SACK uses the optional part of the TCP header. See #TCP segment structure. The use of SACK is widespread - all popular TCP stacks support it. Selective acknowledgment is also used in SCTP. In the field of computer networking, the IETF Signaling Transport (SIGTRAN) working group defined the Stream Control Transmission Protocol (SCTP) as a transport layer protocol in 2000. ...


Error-free data transfer

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 TCP segment structure for details on checksumming). A checksum is a form of redundancy check, a simple way to protect the integrity of data by detecting errors in data that are 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 function that takes as input a data stream of any length and produces as output a value of a certain fixed size. ... 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, diverse family of frame-based computer networking technologies that operate at many speeds 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. ... The end-to-end principle is one of the central design principles of the Transmission Control Protocol (TCP) widely used on the Internet. ...

A Simplified TCP State Diagram. See * TCP EFSM diagram for a more detailed state diagram including the states inside the ESTABLISHED state.
A Simplified TCP State Diagram. See * TCP EFSM diagram for a more detailed state diagram including the states inside the ESTABLISHED state.

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 ...

Congestion control

The final part to TCP is congestion control. TCP uses a number of mechanisms to achieve high performance and avoid 'congestion collapse', where network performance can fall by several orders of magnitude. These mechanisms control the rate of data entering the network, keeping the data flow below a rate that would trigger collapse. Congestion control concerns controlling traffic entry into a telecommunications network, so as to avoid congestive collapse by attempting to avoid oversubscription of any of the processing or link capabilities of the intermediate nodes and networks and taking resource reducing steps, such as reducing the rate of sending packets. ... Congestive collapse (or congestion collapse) is a condition which a packet switched computer network can reach, when little or no useful communication is happening due to congestion. ...


Acknowledgments for data sent, or lack of acknowledgments, 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.


Modern implementations of TCP contain four intertwined algorithms: Slow-start, congestion avoidance, fast retransmit, and fast recovery (RFC2581). 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. ... The TCP uses a network congestion avoidance algorithm that includes various aspects of an additive-increase-multiplicative-decrease (AIMD) scheme, with other schemes such as slow-start in order to achieve congestion avoidance. ... 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.
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 up to that amount of data before it must wait for an acknowledgment and window update from the receiving host. When a receiver advertises the window size of 0, the sender stops sending data and starts the persist timer. The persist timer is used to protect TCP from the dead lock situation. The dead lock situation could be when the new window size update from the receiver is lost and the receiver has no more data to send while the sender is waiting for the new window size update. When the persist timer expires the TCP sender sends a small packet so that the receivers ACKs the packet with the new window size and TCP can recover from such situations. 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 between 2 and 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.


Many routers and packet firewalls rewrite the window scaling factor during a transmission. This causes sending and receiving sides to assume different TCP window sizes. The result is non-stable traffic that is very slow. The problem is visible on some sending and receiving sites which are behind the path of broken routers.


For more information on problems that may be caused, especially with Linux and Vista systems, please see main topic TCP window scale option. The TCP window scale option is an option to increase the TCP congestion window size above its maximum value of 65,536 bytes. ...


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 tear down 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 to terminate the connection by 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.


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.


Some host TCP stacks may implement a "half-duplex" close sequence, as Linux or HP-UX do. If such a host actively closes a connection but still has not read all the incoming data the stack already received from the link, this host will send a RST instead of a FIN (Section 4.2.2.13 in RFC 1122). This allows a TCP application to be sure that the remote application has read all the data the former sent - waiting the FIN from the remote side when it will actively close the connection. Unfortunatelly, the remote TCP stack cannot distinguish between a Connection Aborting RST and this Data Loss RST - both will make the remote stack to throw away all the data it received, but the application still didn't read. This article is about operating systems that use the Linux kernel. ... HP-UX (Hewlett Packard UniX) is Hewlett-Packards proprietary implementation of the Unix operating system, based on System V (initially System III). ...


Some application protocols may violate the OSI model layers, using the TCP open/close handshaking for the application protocol open/close handshaking - these may find the RST problem on active close. As an example: The Open Systems Interconnection Basic 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 (OSI) initiative. ...

 s = connect(remote); send(s, data); close(s); 

For a usual program flow like above, a TCP/IP stack like that described above does not guarantee that all the data will arrive to the other application unless the programmer is sure that the remote side will not send anything.


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). ... An Internet socket (or commonly, a socket or network socket), is a communication end-point unique to a machine communicating on an Internet Protocol-based network, such as the Internet. ...


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), ssh (22), 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. The Internet Assigned Numbers Authority (IANA) is the entity that oversees global IP address allocation, DNS root zone management, and other Internet protocol assignments. ... This article is about the File Transfer Protocol standardised by the IETF. For other file transfer protocols, see File transfer protocol (disambiguation). ... Secure Shell or SSH is a network protocol that allows data to be exchanged over a secure channel between two computers. ... For the packet switched network, see Telenet. ... 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 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 its first specification RFC 675 in 1974, and the v4 specification RFC 793, published in September 1981.[3] 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. The TCP uses a network congestion avoidance algorithm that includes various aspects of an additive-increase-multiplicative-decrease (AIMD) scheme, with other schemes such as slow-start in order to achieve congestion avoidance. ... 1981 is a common year starting on Thursday of the Gregorian calendar. ... 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. ...


The original TCP congestion avoidance algorithm was known as "TCP Tahoe", but many alternative algorithms have since been proposed (including TCP Reno, Vegas, FAST TCP, New Reno, and Hybla). The TCP uses a network congestion avoidance algorithm that includes various aspects 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. ... This does not cite its references or sources. ...


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, hand off, 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). Image File history File links Broom_icon. ... 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 hybrid 2. ...


Hardware TCP implementations

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. Alacritech’s 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. Some networking stacks support the SO_DEBUG socket option, which can be enabled on the socket using setsockopt. That option dumps all the packets, TCP states and events on that socket which will be helpful in debugging. netstat is another utility that can be used for debugging. A packet sniffer (also known as a network analyzer or protocol analyzer or, for particular types of networks, an Ethernet sniffer or wireless sniffer) is computer software or computer hardware that can intercept and log traffic passing over a digital network or part of a network. ... This article or section includes a list of works cited or a list of external links, but its sources remain unclear because it lacks in-text citations. ...


Alternatives to TCP

For many applications TCP is not appropriate. One 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. Web radio (or Internet radio) is a broadcasting service transmitted via the Internet. ... An overview of how VoIP works A typical analog telephone adapter for connecting an ordinary phone to a VoIP network Ciscos implementation of VoIP - IP Phone Voice over Internet Protocol, also called VoIP (pronounced voyp), IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband is the...


Also for embedded systems, network booting and servers that serve simple requests from huge numbers of clients (e.g. DNS servers) the complexity of TCP can be a problem. 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. What is an Embedded System? Electronic devices that incorporate a computer(usually a microprocessor) within their implementation. ... Network booting is the process of booting a computer from a network rather than a local drive. ... On the Internet, the Domain Name Server (DNS) associates various sorts of information with so-called domain names; most importantly, it serves as the phone book for the Internet by translating human-readable computer hostnames, e. ... In Computer Networking, the process of Network Address Translation (NAT, also known as Network Masquerading, Native Address Translation or IP Masquerading) involves re-writing the source and/or destination addresses of IP packets as they pass through a Router or firewall. ... This article is about the Internet protocol. ...


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. 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). ... In telecommunication, forward error correction (FEC) is a system of error control for data transmission, whereby the sender adds redundant data to its messages, which allows the receiver to detect and correct errors (within some bound) without the need to ask the sender for additional data. ... This article does not cite any references or sources. ...


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. In the field of computer networking, the IETF Signaling Transport (SIGTRAN) working group defined the Stream Control Transmission Protocol (SCTP) as a transport layer protocol in 2000. ...


Venturi Transport Protocol (VTP) is a patented proprietary protocol that is designed to replace TCP transparently in order to overcome perceived inefficiencies related to wireless data transport. Venturi Transport Protocol (VTP) is a patented proprietary transport layer protocol that is designed to transparently replace TCP in order to overcome perceived inefficiencies related to wireless data transport. ...


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 the retransmits that TCP needs. The TCP uses a network congestion avoidance algorithm that includes various aspects 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, packet switching network and data link layer protocol which encodes data traffic into small (53 bytes; 48 bytes of data and 5 bytes of header information) fixed-sized cells. ...


Fields used to compute the checksum

TCP checksum using IPv4

When TCP runs over IPv4, the method used to compute the checksum is defined in RFC 793: 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. ...

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 IPv6. The difference is in the data used to make the checksum. When computing the checksum, a pseudo-header that mimics the IPv4 header is shown in the table below. In mathematics, signed numbers in some arbitrary base is done in the usual way, by prefixing it with a - sign. ... Internet Protocol version 6 (IPv6) is a network layer protocol for packet-switched internetworks. ...

TCP pseudo-header (IPv4)
Bit offset Bits 0–3 4–7 8–15 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. This is a list of IP protocol numbers that defines the number used in the protocol field of IPv4 packets. ...


TCP checksum using IPv6

When TCP runs over IPv6, the method used to compute the checksum is changed, as per RFC 2460: Internet Protocol version 6 (IPv6) is a network layer protocol for packet-switched internetworks. ...

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 shown in the table below.

TCP pseudo-header (IPv6)
Bit offset Bits 0 - 7 8–15 16–23 24–31
0 Source address
32
64
96
128 Destination address
160
192
224
256 TCP length
288 Zeros Next header
320 Source port Destination port
352 Sequence number
384 Acknowledgement number
416 Data offset Reserved Flags Window
448 Checksum Urgent pointer
480 Options (optional)
480/512+  
Data
 
  • Source address – the one in the IPv6 header
  • Destination address – 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.
  • TCP length – the length of the TCP header and data;
  • Next Header – the protocol value for TCP

See also

A connection-oriented networking protocol is one which identifies traffic flows by some connection identifier rather than by explicitly listing source and destination addresses. ... T/TCP (Transactional TCP) is a variant of the TCP protocol. ... 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. ... The TCP uses various variations of an additive-increase-multiplicative-decrease (AIMD) scheme, with other schemas such as slow-start in order to achieve congestion avoidance. ... Transmission Control Protocol (TCP) accepts data from a data stream, segments it into chunks, and adds a TCP header creating a The tcp segment is then encapsulated, into an IP datagram. ... There are very few or no other articles that link to this one. ... To meet Wikipedias quality standards, this article or section may require cleanup. ... 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. ... A normal connection between a user (Alice) and a server. ... There are very few or no other articles that link to this one. ... C is a general-purpose, block structured, procedural, imperative computer programming language developed in 1972 by Dennis Ritchie at the Bell Telephone Laboratories for use with the Unix operating system. ... In the field of computer networking, the IETF Signaling Transport (SIGTRAN) working group defined the Stream Control Transmission Protocol (SCTP) as a transport layer protocol in 2000. ... In computing and telecommunications, the transport layer is the second highest layer in the four and five layer TCP/IP reference models, where it responds to service requests from the application layer and issues service requests to the Internet layer. ...

References

  1. ^ Comer, Douglas E. (2006). Internetworking with TCP/IP:Principles, Protocols, and Architecture, 5th, Prentice Hall. 
  2. ^ Laurent Joncheray, Simple Active Attack Against TCP, 1995 [1]

External links


  Results from FactBites:
 
Transmission Control Protocol - Wikipedia, the free encyclopedia (3273 words)
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.
TCP window scale, as defined in RFC 1323, is an option used to increase the maximum window size from 65,535 bytes to 1 Gigabyte.
The TCP length field is the length of the TCP header and data.
The Transmission Control Protocol (3799 words)
TCP performance is often dependent on a subset of algorithms and techniques such as flow control and congestion control.
All TCP segments carry a checksum, which is used by the receiver to detect errors with either the TCP header or data.
Flow control is a technique whose primary purpose is to properly match the transmission rate of sender to that of the receiver and the network.
  More results at FactBites »

 
 

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