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).
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.
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.
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).
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 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.
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 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.
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.
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.
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: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.
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.
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.[1] 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. Common applications that use TCP include HTTP (World Wide Web), SMTP (e-mail) and FTP (file transfer).
The original TCP congestion control was called TCP Tahoe, several alternative congestion control algorithms have been proposed:
BIC TCP by Lisong Xu, Khaled Harfoush, and Injong Rhee at North Carolina State University
Compound TCP by K. Tan, J. Song, Q. Zhang, and M. Sridharan at Microsoft Research
CUBIC by Injong Rhee, and Lisong Xu
Fast TCP by Cheng Jin, David X. Wei and Steven H. Low. at Caltech.
H-TCP by D. Leithi, and R. Shorten at Hamilton Institute
High Speed TCP proposed by S. Floyd in RFC 3649
HSTCP-LP by A. Kuzmanovic, E. W. Knightly, and R. Les Cottrell
NewReno, proposed by S. Floyd, T. Henderson and A. Gurtov in RFC 3782
Scalable TCP by Tom Kelly
TCP Hybla by Carlo Caini and Rosario Firrincieli at University of Bologna
TCP-Illinois by Shao Liu, Tamer Basar and R. Srikant
TCP-LP by Aleksandar Kuzmanovic
TCP Reno by BSD 4.3BSD
TCP SACK
TCP Vegas by Lawrence S. Brakmo and Larry L. Peterson at University of Arizona
TCP Veno by C. P. Fu, S. C. Liew
TCP Westwood by Saverio Mascolo, Claudio Casetti, Mario Gerla, M. Y. Sanadidi, and Ren Wang
TCP Westwood+ by A. Dell’Aera, L. A. Grieco, S. Mascolo
XCP by Aaron Falk, Dina Katabi
YeAH-TCP by Andrea Baiocchi, Angelo P. Castellani and Francesco Vacirca.
Ensemble Flow Congestion Management (EFCM), Fuzzy Explicit Window Adaptation (FEWA), Enhanced TCP (ETCP)[2]
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.
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).
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.
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.
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).
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 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.
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 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.
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.
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.
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: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.
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.
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.[1] 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. Common applications that use TCP include HTTP (World Wide Web), SMTP (e-mail) and FTP (file transfer).
The original TCP congestion control was called TCP Tahoe, several alternative congestion control algorithms have been proposed:
BIC TCP by Lisong Xu, Khaled Harfoush, and Injong Rhee at North Carolina State University
Compound TCP by K. Tan, J. Song, Q. Zhang, and M. Sridharan at Microsoft Research
CUBIC by Injong Rhee, and Lisong Xu
Fast TCP by Cheng Jin, David X. Wei and Steven H. Low. at Caltech.
H-TCP by D. Leithi, and R. Shorten at Hamilton Institute
High Speed TCP proposed by S. Floyd in RFC 3649
HSTCP-LP by A. Kuzmanovic, E. W. Knightly, and R. Les Cottrell
NewReno, proposed by S. Floyd, T. Henderson and A. Gurtov in RFC 3782
Scalable TCP by Tom Kelly
TCP Hybla by Carlo Caini and Rosario Firrincieli at University of Bologna
TCP-Illinois by Shao Liu, Tamer Basar and R. Srikant
TCP-LP by Aleksandar Kuzmanovic
TCP Reno by BSD 4.3BSD
TCP SACK
TCP Vegas by Lawrence S. Brakmo and Larry L. Peterson at University of Arizona
TCP Veno by C. P. Fu, S. C. Liew
TCP Westwood by Saverio Mascolo, Claudio Casetti, Mario Gerla, M. Y. Sanadidi, and Ren Wang
TCP Westwood+ by A. Dell’Aera, L. A. Grieco, S. Mascolo
XCP by Aaron Falk, Dina Katabi
YeAH-TCP by Andrea Baiocchi, Angelo P. Castellani and Francesco Vacirca.
Ensemble Flow Congestion Management (EFCM), Fuzzy Explicit Window Adaptation (FEWA), Enhanced TCP (ETCP)[2]
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.
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).
