Transmission Control Protocol (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 and congestion window to achieve congestion avoidance. The TCP congestion-avoidance algorithm is the primary basis for congestion control in the Internet. Per the end-to-end principle, congestion control is largely a function of internet hosts, not the network itself. There are several variations and versions of the algorithm implemented in protocol stacks of operating systems of computers that connect to the Internet.
To avoid congestive collapse, TCP uses a multi-faceted congestion-control strategy. For each connection, TCP maintains a congestion window, limiting the total number of unacknowledged packets that may be in transit end-to-end. This is somewhat analogous to TCP's sliding window used for flow control. TCP uses a mechanism called slow start to increase the congestion window after a connection is initialized or after a timeout. It starts with a window a small multiple of the maximum segment size (MSS) in size. Although the initial rate is low, the rate of increase is very rapid; for every packet acknowledged, the congestion window increases by 1 MSS so that the congestion window effectively doubles for every round-trip time (RTT).
When the congestion window exceeds the slow-start threshold, ssthresh,[a] the algorithm enters a new state, called congestion avoidance. In congestion avoidance state, as long as non-duplicate ACKs are received[b] the congestion window is additively increased by one MSS every round-trip time.
In TCP, the congestion window is one of the factors that determines the number of bytes that can be outstanding at any time. The congestion window is maintained by the sender. Note that this is not to be confused with the sliding window size which is maintained by the receiver. The congestion window is a means of stopping a link between the sender and the receiver from becoming overloaded with too much traffic. It is calculated by estimating how much congestion there is on the link.
When a connection is set up, the congestion window, a value maintained independently at each host, is set to a small multiple of the MSS allowed on that connection. Further variance in the congestion window is dictated by an AIMD approach. This means that if all segments are received and the acknowledgments reach the sender on time, some constant is added to the window size. When the window reaches ssthresh. Once the sender reaches this threshold, the congestion window increases linearly at the rate of 1/(congestion window) segment on each new acknowledgement received. The window keeps growing until a timeout occurs. On timeout:
The flow of data over a TCP connection is also controlled by the use of the receive window advertised by the receiver. By comparing its own congestion window with the receive window, a sender can determine how much data it may send at any given time.
Slow-start is part of the congestion control strategy used by TCP, the data transmission protocol used by many Internet applications. Slow-start is used in conjunction with other algorithms to avoid sending more data than the network is capable of transmitting, that is, to avoid causing network congestion. The algorithm is specified by RFC 5681.
Slow-start begins initially with a congestion window size (cwnd) of 1, 2, 4 or 10 MSS.:1 The value of the Congestion Window will be increased by one with each acknowledgement (ACK) received, effectively doubling the window size each round-trip time ("although it is not exactly exponential because the receiver may delay its ACKs, typically sending one ACK for every two segments that it receives"). The transmission rate will be increased with slow-start algorithm until either a loss is detected, or the receiver's advertised window (rwnd) is the limiting factor, or ssthresh is reached. If a loss event occurs, TCP assumes that it is due to network congestion and takes steps to reduce the offered load on the network. These measurements depend on the used TCP congestion avoidance algorithm. Once ssthresh is reached, TCP changes from slow-start algorithm to the linear growth (congestion avoidance) algorithm. At this point, the window is increased by 1 segment for each round-trip delay time (RTT).
Although the strategy is referred to as "Slow-Start", its congestion window growth is quite aggressive, more aggressive than the congestion avoidance phase. Before slow-start was introduced in TCP, the initial pre-congestion avoidance phase was even faster.
The behavior upon packet loss depends on the TCP congestion avoidance algorithm that is used.
Slow start assumes that unacknowledged segments are due to network congestion. While this is an acceptable assumption for many networks, segments may be lost for other reasons, such as poor data link layer transmission quality. Thus, slow start can perform poorly in situations with poor reception, such as wireless networks.
The slow start protocol performs badly for short-lived connections. Older web browsers would create many consecutive short-lived connections to the web server, and would open and close the connection for each file requested. This kept most connections in the slow start mode, which resulted in poor response time. To avoid this problem, modern browsers either open multiple connections simultaneously or reuse one connection for all files requested from a particular web server. However, connections cannot be reused for the multiple third-party servers used by web sites to implement web advertising, sharing features of social networking services, and counter scripts of web analytics.
The additive increase/multiplicative decrease (AIMD) algorithm is a feedback control algorithm. AIMD combines linear growth of the congestion window with an exponential reduction when a congestion takes place. Multiple flows using AIMD congestion control will eventually converge to use equal amounts of a contended link.
A TCP sender uses a timer to recognize lost segments. If an acknowledgement is not received for a particular segment within a specified time (a function of the estimated round-trip delay time), the sender will assume the segment was lost in the network, and will retransmit the segment.
Duplicate acknowledgement is the basis for the fast retransmit mechanism which works as follows: after receiving a packet (e.g. with sequence number 1), the receiver sends an acknowledgement by adding 1 to the sequence number (i.e. acknowledgement number 2) which means that the receiver received the packet number 1 and it expects packet number 2 from the sender. Suppose that three subsequent packets have been lost. In the meantime the receiver receives packet numbers 5 and 6. After receiving packet number 5, the receiver sends an acknowledgement, but still only for sequence number 2. When the receiver receives packet number 6, it sends yet another acknowledgement value of 2. Because the sender receives more than one acknowledgement with the same sequence number (2 in this example) this is called duplicate acknowledgement.
The fast retransmit enhancement works as follows: if a TCP sender receives a specified number of acknowledgements which is usually set to three duplicate acknowledgements with the same acknowledge number (that is, a total of four acknowledgements with the same acknowledgement number), the sender can be reasonably confident that the segment with the next higher sequence number was dropped, and will not arrive out of order. The sender will then retransmit the packet that was presumed dropped before waiting for its timeout.
The "TCP Foo" names for the algorithms appear to have originated in a 1996 paper by Kevin Fall and Sally Floyd.
The following is one possible classification according to the following properties:
Some well-known congestion avoidance mechanisms are classified by this scheme as follows:
|High Speed||Loss||Sender||High bandwidth|
|Compound TCP||Loss/Delay||Sender||High bandwidth||Proportional|
|CLAMP||Multi-bit signal||Receiver, Routers||V||Max-min|
|TFRC||Loss||Sender, Receiver||No Retransmission||Minimum delay|
|XCP||Multi-bit signal||Sender, Receiver, Router||BLFC||Max-min|
|VCP||2-bit signal||Sender, Receiver, Router||BLF||Proportional|
|MaxNet||Multi-bit signal||Sender, Receiver, Router||BLFSC||Max-min|
|JetMax||Multi-bit signal||Sender, Receiver, Router||High bandwidth||Max-min|
|ECN||Single-bit signal||Sender, Receiver, Router||Less loss|
The two algorithms were retrospectively named after the 4.3BSD operating system in June 1986 in which each first appeared (which were themselves named after Lake Tahoe and the nearby city of Reno, Nevada). The "Tahoe" algorithm first appeared in 4.3BSD-Tahoe (which was made to support the CCI Power 6/32 "Tahoe" minicomputer), and was made available to non-AT&T licensees as part of the 4.3BSD Networking Release 1; this ensured its wide distribution and implementation. Improvements were made in 4.3BSD-Reno and subsequently released to the public as Networking Release 2 and later 4.4BSD-Lite.
While both consider retransmission timeout (RTO) and duplicate ACKs as packet loss events, the behavior of Tahoe and Reno differ primarily in how they react to duplicate ACKs:
In both Tahoe and Reno, if an ACK times out (RTO timeout), slow start is used, and both algorithms reduce congestion window to 1 MSS.
In this state, TCP retransmits the missing packet that was signaled by three duplicate ACKs, and waits for an acknowledgment of the entire transmit window before returning to congestion avoidance. If there is no acknowledgment, TCP Reno experiences a timeout and enters the slow start state.
Until the mid-1990s, all of TCP's set timeouts and measured round-trip delays were based upon only the last transmitted packet in the transmit buffer. University of Arizona researchers Larry Peterson and Lawrence Brakmo introduced TCP Vegas, named after the largest Nevada city, in which timeouts were set and round-trip delays were measured for every packet in the transmit buffer. In addition, TCP Vegas uses additive increases in the congestion window. This variant was not widely deployed outside Peterson's laboratory. In a comparison study of various TCP congestion control algorithms, TCP Vegas appeared to be the smoothest followed by TCP CUBIC.
TCP New Reno, defined by RFC 6582 (which obsoletes previous definitions in RFC 3782 and RFC 2582), improves retransmission during the fast-recovery phase of TCP Reno. During fast recovery, for every duplicate ACK that is returned to TCP New Reno, a new unsent packet from the end of the congestion window is sent, to keep the transmit window full. For every ACK that makes partial progress in the sequence space, the sender assumes that the ACK points to a new hole, and the next packet beyond the ACKed sequence number is sent.
Because the timeout timer is reset whenever there is progress in the transmit buffer, this allows New Reno to fill large holes, or multiple holes, in the sequence space - much like TCP SACK. Because New Reno can send new packets at the end of the congestion window during fast recovery, high throughput is maintained during the hole-filling process, even when there are multiple holes, of multiple packets each. When TCP enters fast recovery it records the highest outstanding unacknowledged packet sequence number. When this sequence number is acknowledged, TCP returns to the congestion avoidance state.
A problem occurs with New Reno when there are no packet losses but instead, packets are reordered by more than 3 packet sequence numbers. When this happens, New Reno mistakenly enters fast recovery, but when the reordered packet is delivered, ACK sequence-number progress occurs and from there until the end of fast recovery, every bit of sequence-number progress produces a duplicate and needless retransmission that is immediately ACKed.
New Reno performs as well as SACK at low packet error rates, and substantially outperforms Reno at high error rates.
TCP Hybla aims to eliminate penalization of TCP connections that incorporate a high-latency terrestrial or satellite radio link, due to their longer round-trip times. It stems from an analytical evaluation of the congestion window dynamics, which suggests the necessary modifications to remove the performance dependence on RTT.
Binary Increase Congestion control is an implementation of TCP with an optimized congestion control algorithm for high speed networks with high latency (called LFN, long fat networks, in RFC 1072). BIC is used by default in Linux kernels 2.6.8 through 2.6.18.
CUBIC is a less aggressive and more systematic derivative of BIC, in which the window is a cubic function of time since the last congestion event, with the inflection point set to the window prior to the event. CUBIC is used by default in Linux kernels between versions 2.6.19 and 3.2.
Agile-SD is a Linux-based Congestion Control Algorithm (CCA) which is designed for the real Linux kernel. It is a receiver-side algorithm employs a loss-based approach using a novel mechanism, called Agility Factor (AF). It has been proposed by Mohamed A. Alrshah et al. to increase the bandwidth utilization over high-speed and short-distance networks (low-BDP networks) such as local area networks or fiber-optic network, especially when the applied buffer size is small. It has been evaluated by comparing its performance to Compound-TCP (the default CCA in MS Windows) and CUBIC (the default of Linux) using NS-2 simulator. It improves the total performance up to 55% in term of average throughput.
Westwood+ is a sender-side only modification of the TCP Reno protocol stack that optimizes the performance of TCP congestion control over both wireline and wireless networks. TCP Westwood+ is based on end-to-end bandwidth estimation to set congestion window and slow start threshold after a congestion episode, that is, after three duplicate acknowledgments or a timeout. The bandwidth is estimated by properly low-pass filtering the rate of returning acknowledgment packets. The rationale of this strategy is simple: in contrast with TCP Reno, which blindly halves the congestion window after three duplicate ACKs, TCP Westwood+ adaptively sets a slow start threshold and a congestion window which takes into account the bandwidth used at the time congestion is experienced. TCP Westwood+ significantly increases throughput over wireless links and fairness compared to TCP Reno/New Reno in wired networks.
Compound TCP is a Microsoft implementation of TCP which maintains two different congestion windows simultaneously, with the goal of achieving good performance on LFNs while not impairing fairness. It has been widely deployed in Windows versions since Microsoft Windows Vista and Windows Server 2008 and has been ported to older Microsoft Windows versions as well as Linux.
TCP Proportional Rate Reduction (PRR) is an algorithm designed to improve the accuracy of data sent during recovery. The algorithm ensures that the window size after recovery is as close as possible to the slow start threshold. In tests performed by Google, PRR resulted in a 3-10% reduction in average latency and recovery timeouts reduced by 5%. PRR is used by default in Linux kernels since version 3.2.
Bottleneck Bandwidth and Round-trip propagation time (BBR) is a TCP congestion control algorithm developed at Google in 2016. While most congestion control algorithms are loss-based, in that they rely on packet loss as a signal to lower rates of transmission, BBR is model-based. The algorithm uses the maximum bandwidth and round-trip time at which the network delivered the most recent flight of outbound data packets to build an explicit model of the network. Each cumulative or selective acknowledgment of packet delivery produces a rate sample which records the amount of data delivered over the time interval between the transmission of a data packet and the acknowledgment of that packet. As network interface controllers evolve from megabit per second to gigabit per second performance,[clarification needed] packet loss should no longer be considered the primary determining factor in identifying congestion, making model-based congestion control algorithms which provide higher throughput and lower latency, such as BBR, a more reliable alternative to more popular algorithms like CUBIC.
TCP New Reno was the most commonly implemented algorithm, SACK support is very common and is an extension to Reno/New Reno. Most others are competing proposals which still need evaluation. Starting with 2.6.8 the Linux kernel switched the default implementation from New Reno to BIC. The default implementation was again changed to CUBIC in the 2.6.19 version. FreeBSD uses New Reno as the default algorithm. However, it supports a number of other choices.
When the per-flow product of bandwidth and latency increases, regardless of the queuing scheme, TCP becomes inefficient and prone to instability. This becomes increasingly important as the Internet evolves to incorporate very high-bandwidth optical links.
TCP Interactive (iTCP) allows applications to subscribe to TCP events and respond accordingly enabling various functional extensions to TCP from outside TCP layer. Most TCP congestion schemes work internally. iTCP additionally enables advanced applications to directly participate in congestion control such as to control the source generation rate.
Zeta-TCP detects the congestions from both the latency and loss rate measures, and applies different congestion window backoff strategies based on the likelihood of the congestions to maximize the goodput. It also has a couple of other improvements to accurately detect the packet losses, avoiding retransmission timeout retransmission; and accelerate/control the inbound (download) traffic.
Congestion control algorithms are classified in relation to network awareness, meaning the extent to which these algorithms are aware of the state of the network, and consist of three primary categories: black box, grey box, and green box.
Black box algorithms offer blind methods of congestion control. They operate only on the binary feedback received upon congestion and do not assume any knowledge concerning the state of the networks which they manage.
Grey box algorithms use time-instances in order to obtain measurements and estimations of bandwidth, flow contention, and other knowledge of network conditions.
Green box algorithms offer bimodal methods of congestion control which measures the fair-share of total bandwidth which should be allocated for each flow, at any point, during the system's execution.
The following algorithms require custom fields to be added to the TCP packet structure:
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