TCP congestion control
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, 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:
- Congestion window is reset to 1 MSS.
- ssthresh is set to half the congestion window size before the timeout.
- slow start is initiated.
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 in conjunction with other algorithms to avoid sending more data than the network is capable of forwarding, that is, to avoid causing network congestion. The algorithm is specified by RFC 5681.
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.
Slow start begins initially with a congestion window size (cwnd) of 1, 2, 4 or 10 MSS.:1 The value for the congestion window size will be increased by one with each acknowledgement (ACK) received, effectively doubling the window size each round-trip time.[c] The transmission rate will be increased by the 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 exact TCP congestion avoidance algorithm used.
- TCP Tahoe
- When a loss occurs, fast retransmit is sent, half of the current CWND is saved as ssthresh and slow start begins again from its initial CWND. Once the CWND reaches ssthresh, TCP changes to congestion avoidance algorithm where each new ACK increases the CWND by MSS / CWND. This results in a linear increase of the CWND.
- TCP Reno
- A fast retransmit is sent, half of the current CWND is saved as ssthresh and as new CWND, thus skipping slow start and going directly to the congestion avoidance algorithm. The overall algorithm here is called fast recovery.
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).
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 also 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. Connections, however, 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.
Additive increase/multiplicative decreaseEdit
The additive increase/multiplicative decrease (AIMD) algorithm is a closed-loop 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.
Fast retransmit is an enhancement to TCP that reduces the time a sender waits before retransmitting a lost segment. A TCP sender normally uses a simple 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. 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). This indicates to the sender that the receiver received the packet number 1 and it expects packet number 2. Suppose that three subsequent packets are lost. The next packets the receiver sees are 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. Duplicate acknowledgement occurs when the sender receives more than one acknowledgement with the same sequence number (2 in this example).
When a sender receives several duplicate acknowledgements, it can be reasonably confident that the segment with the next higher sequence number was dropped. A sender with fast retransmit will then retransmit this packet immediately without waiting for its timeout.
The following is one possible classification according to the following properties:
- the type and amount of feedback received from the network
- incremental deployability on the current Internet
- the aspect of performance it aims to improve: high bandwidth-delay product networks (B); lossy links (L); fairness (F); advantage to short flows (S); variable-rate links (V); speed of convergence (C)
- the fairness criterion it uses
Some well-known congestion avoidance mechanisms are classified by this scheme as follows:
|High Speed||Loss||Sender||High bandwidth|
|C2TCP||Loss/Delay||Sender||Ultra-low latency and high bandwidth|
|Elastic-TCP||Loss and Delay||Sender||High bandwidth/short & long-distance|
|Compound TCP||Loss/Delay||Sender||High bandwidth||Proportional|
|CLAMP||Multi-bit signal||Receiver, Router||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||Reduced loss|
TCP Tahoe and RenoEdit
TCP Tahoe and Reno 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:
- Tahoe: If three duplicate ACKs are received (i.e. four ACKs acknowledging the same packet, which are not piggybacked on data and do not change the receiver's advertised window), Tahoe performs a fast retransmit, sets the slow start threshold to half of the current congestion window, reduces the congestion window to 1 MSS, and resets to slow start state.
- Reno: If three duplicate ACKs are received, Reno will perform a fast retransmit and skip the slow start phase by instead halving the congestion window (instead of setting it to 1 MSS like Tahoe), setting the slow start threshold equal to the new congestion window, and enter a phase called fast recovery.
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.
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 RenoEdit
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 ReductionEdit
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 available 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.
According to Geoff Huston  BBR is efficient and fast, but highly unfair to other, non-BBR streams. He calls it "not scalable" and says it "has the ability to slaughter concurrent loss-based flows". Hock, Bless and Zitterbart  evaluated BBR's implementation in Linux 4.9. If used with one flow, they confirmed the claimed features, but also found "some severe inherent issues such as increased queuing delays, unfairness, and massive packet loss".
Moreover, Soheil Abbasloo et al. in, and  show that BBR doesn't perform well in dynamic environments such as cellular networks. They have also shown that BBR has serious TCP-unfriendliness issue. In other words, for example, when TCP Cubic (which is the default TCP algorithm in Linux, Android, MacOS, etc.) coexists with a BBR flow in the network, BBR flow can simply kill Cubic flow and get the whole link bandwidth from it (e.g. check figure 18 in ).
Elastic-TCP has been proposed in February 2019 by Mohamed A. Alrshah et al. to increase the bandwidth utilization over high-BDP networks to support recent applications such as cloud computing, big data transfer, IoT, etc. It is a Linux-based Congestion Control Algorithm (CCA) which is designed for the Linux kernel. It is a receiver-side algorithm that employs a Loss-delay-based approach using a novel mechanism, called Window-correlated Weighting Function (WWF). It has a high level of elasticity to deal with different network characteristics without the need for human tuning. It has been evaluated by comparing its performance to Compound-TCP (the default CCA in MS Windows), CUBIC (the default of Linux) and TCP-BBR (the default of Linux 4.9 by Google) using NS-2 simulator and testbed. Elastic-TCP significantly improves the total performance in term of average throughput, loss ratio, and delay.
Cellular Controlled Delay TCP (C2TCP) was motivated by the lack of a flexible end-to-end TCP approach that can satisfy various QoS requirements of different applications without requiring any changes in the network devices. C2TCP aims to satisfy ultra-low latency and high bandwidth requirements of applications such as virtual reality, video conferencing, online gaming, vehicular communication systems, etc. in a highly dynamic environment such as current LTE and future 5G cellular networks. C2TCP works as an add-on on top of loss-based TCP (e.g. Reno, NewReno, Cubic, BIC, ...) and makes the average delay of packets bounded to the desired delays set by the applications.
Researchers at NYU showed that C2TCP outperforms the delay/Jitter performance of various state-of-the-art TCP schemes. For instance, they showed that compared to BBR, Cubic, and Westwood on average, C2TCP decreases the average delay of packets by about 250%, 900%, and 700% respectively on various cellular network environments.
C2TCP is only required to be installed on the server-side.
Other TCP congestion avoidance algorithmsEdit
- FAST TCP
- Generalized FAST TCP
- Data Center TCP
- High Speed TCP
- TCP SACK
- Scalable TCP
- TCP Veno
- Congestion Avoidance with Normalized Interval of Time (CANIT)
- Non-linear neural network congestion control based on genetic algorithm for TCP/IP networks
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.
Classification by network awarenessEdit
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.
- BIC TCP (Binary Increase Congestion Control Protocol) uses a concave increase of the sources rate after each congestion event until the window is equal to that before the event, in order to maximise the time that the network is fully utilised. After that, it probes aggressively.
- CUBIC TCP – 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.
- AIMD-FC (Additive Increase Multiplicative Decrease with Fast Convergence), an improvement of AIMD.
- Binomial Mechanisms
- SIMD Protocol
- TCP Vegas – estimates the queuing delay, and linearly increases or decreases the window so that a constant number of packets per flow are queued in the network. Vegas implements proportional fairness.
- FAST TCP – achieves the same equilibrium as Vegas, but uses proportional control instead of linear increase, and intentionally scales the gain down as the bandwidth increases with the aim of ensuring stability.
- TCP-Westwood (TCPW) – a loss causes the window to be reset to the sender's estimate of the bandwidth-delay product, which is the smallest measured RTT times the observed rate of receiving ACKs.
- Bimodal Mechanism – Bimodal Congestion Avoidance and Control mechanism.
- Signalling methods implemented by routers
- Network-Assisted Congestion Control
The following algorithms require custom fields to be added to the TCP packet structure:
- Explicit Control Protocol (XCP) – XCP routers signal explicit increase and decreases in the senders' congestion windows.
- MaxNet – MaxNet uses a single header field, which carries the maximum congestion level of any router on a flow's path. The rate is set as a function of this maximum congestion, resulting in max-min fairness.
- JetMax – JetMax, like MaxNet, also responds only to the maximum congestion signal, but also carries other overhead fields
- BIC is used by default in Linux kernels 2.6.8 through 2.6.18. (August 2004 – September 2006)
- CUBIC is used by default in Linux kernels since version 2.6.19. (November 2006)
- PRR is incorporated in Linux kernels to improve loss recovery since version 3.2. (January 2012)
- BBR is incorporated in Linux kernels to enable model-based congestion control since version 4.9. (December 2016)
- In some implementations (e.g., Linux), the initial ssthresh is large, and so the first slow start usually ends after a loss. However, ssthresh is updated at the end of each slow start, and will often affect subsequent slow starts triggered by timeouts.
- When a packet is lost, the likelihood of duplicate ACKs being received is very high. It is also possible in this case, though unlikely, that the stream just underwent extreme packet reordering, which would also prompt duplicate ACKs.
- Even if, actually, the receiver may delay its ACKs, typically sending one ACK for every two segments that it receives
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