External links
 | This computer science article is a stub. You can help Wikipedia by expanding it. |
Type of Transmission Control Protocol which is designed to provide much higher throughput and scalability.
Standard TCP recommendations as per RFC 2581 and RFC 5681 call for congestion window to be halved for each packet lost. Effectively, this process keeps halving the throughput until packet loss stops. Once the packet loss subsides, slow start kicks in to ramp the speed back up. When the window sizes are small, say 1 Mbit/s @ 200 ms round trip time and the window is about 20 packets, this recovery time is quite fast—on the order of a few seconds. But as transfer speeds approach 1 Gbit/s, the recovery time becomes half an hour and for 10 Gbit/s it's over 4 hours. [1] [2]
Scalable TCP modifies the congestion control algorithm. Instead of halving the congestion window size, each packet loss decreases the congestion window by a small fraction (a factor of 1/8 instead of Standard TCP's 1/2) until packet loss stops. When packet loss stops, the rate is ramped up at a slow fixed rate (one packet is added for every one hundred successful acknowledgements) instead of the Standard TCP rate that's the inverse of the congestion window size (thus very large windows take a long time to recover). This helps reduce the recovery time on 10 Gbit/s links from 4+ hours (using Standard TCP) to less than 15 seconds when the round trip time is 200 milliseconds. [1]
The Transmission Control Protocol (TCP) is one of the main protocols of the Internet protocol suite. It originated in the initial network implementation in which it complemented the Internet Protocol (IP). Therefore, the entire suite is commonly referred to as TCP/IP. TCP provides reliable, ordered, and error-checked delivery of a stream of octets (bytes) between applications running on hosts communicating via an IP network. Major internet applications such as the World Wide Web, email, remote administration, and file transfer rely on TCP, which is part of the Transport layer of the TCP/IP suite. SSL/TLS often runs on top of TCP.
Network throughput refers to the rate of message delivery over a communication channel, such as Ethernet or packet radio, in a communication network. The data that these messages contain may be delivered over physical or logical links, or through network nodes. Throughput is usually measured in bits per second, and sometimes in data packets per second or data packets per time slot.
Explicit Congestion Notification (ECN) is an extension to the Internet Protocol and to the Transmission Control Protocol and is defined in RFC 3168 (2001). ECN allows end-to-end notification of network congestion without dropping packets. ECN is an optional feature that may be used between two ECN-enabled endpoints when the underlying network infrastructure also supports it.
Network congestion in data networking and queueing theory is the reduced quality of service that occurs when a network node or link is carrying more data than it can handle. Typical effects include queueing delay, packet loss or the blocking of new connections. A consequence of congestion is that an incremental increase in offered load leads either only to a small increase or even a decrease in network throughput.
FAST TCP is a TCP congestion avoidance algorithm especially targeted at long-distance, high latency links, developed at the Netlab, California Institute of Technology and now being commercialized by FastSoft. FastSoft was acquired by Akamai Technologies in 2012.
Throughput of a network can be measured using various tools available on different platforms. This page explains the theory behind what these tools set out to measure and the issues regarding these measurements.
Transmission Control Protocol (TCP) uses a congestion control algorithm that includes various aspects of an additive increase/multiplicative decrease (AIMD) scheme, along with other schemes including slow start and congestion window (CWND), 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.
TCP tuning techniques adjust the network congestion avoidance parameters of Transmission Control Protocol (TCP) connections over high-bandwidth, high-latency networks. Well-tuned networks can perform up to 10 times faster in some cases. However, blindly following instructions without understanding their real consequences can hurt performance as well.
Packet loss occurs when one or more packets of data travelling across a computer network fail to reach their destination. Packet loss is either caused by errors in data transmission, typically across wireless networks, or network congestion. Packet loss is measured as a percentage of packets lost with respect to packets sent.
In data communications, the bandwidth-delay product is the product of a data link's capacity and its round-trip delay time. The result, an amount of data measured in bits, is equivalent to the maximum amount of data on the network circuit at any given time, i.e., data that has been transmitted but not yet acknowledged. The bandwidth-delay product was originally proposed as a rule of thumb for sizing router buffers in conjunction with congestion avoidance algorithm random early detection (RED).
The additive-increase/multiplicative-decrease (AIMD) algorithm is a feedback control algorithm best known for its use in TCP congestion control. AIMD combines linear growth of the congestion window when there is no congestion with an exponential reduction when congestion is detected. Multiple flows using AIMD congestion control will eventually converge to an equal usage of a shared link. The related schemes of multiplicative-increase/multiplicative-decrease (MIMD) and additive-increase/additive-decrease (AIAD) do not reach stability.
In computer networks, goodput is the application-level throughput of a communication; i.e. the number of useful information bits delivered by the network to a certain destination per unit of time. The amount of data considered excludes protocol overhead bits as well as retransmitted data packets. This is related to the amount of time from the first bit of the first packet sent until the last bit of the last packet is delivered.
WAN optimization is a collection of techniques for improving data transfer across wide area networks (WANs). In 2008, the WAN optimization market was estimated to be $1 billion, and was to grow to $4.4 billion by 2014 according to Gartner, a technology research firm. In 2015 Gartner estimated the WAN optimization market to be a $1.1 billion market.
The TCP window scale option is an option to increase the receive window size allowed in Transmission Control Protocol above its former maximum value of 65,535 bytes. This TCP option, along with several others, is defined in RFC 7323 which deals with long fat networks (LFNs).
In computing, Microsoft's Windows Vista and Windows Server 2008 introduced in 2007/2008 a new networking stack named Next Generation TCP/IP stack, to improve on the previous stack in several ways. The stack includes native implementation of IPv6, as well as a complete overhaul of IPv4. The new TCP/IP stack uses a new method to store configuration settings that enables more dynamic control and does not require a computer restart after a change in settings. The new stack, implemented as a dual-stack model, depends on a strong host-model and features an infrastructure to enable more modular components that one can dynamically insert and remove.
A sliding window protocol is a feature of packet-based data transmission protocols. Sliding window protocols are used where reliable in-order delivery of packets is required, such as in the data link layer as well as in the Transmission Control Protocol (TCP). They are also used to improve efficiency when the channel may include high latency.
TCP-Friendly Rate Control (TFRC) is a congestion control mechanism designed for unicast flows operating in an Internet environment and competing with TCP traffic. The goal is to compete fairly with TCP traffic on medium timescales, but to be much less variable than TCP on short timescales.
Bufferbloat is a cause of high latency and jitter in packet-switched networks caused by excess buffering of packets. Bufferbloat can also cause packet delay variation, as well as reduce the overall network throughput. When a router or switch is configured to use excessively large buffers, even very high-speed networks can become practically unusable for many interactive applications like voice over IP (VoIP), audio streaming, online gaming, and even ordinary web browsing.
Zeta-TCP refers to a set of proprietary Transmission Control Protocol (TCP) algorithms aiming at improving the end-to-end performance of TCP, regardless of whether the peer is Zeta-TCP or any other TCP protocol stack, in other words, to be compatible with the existing TCP algorithms. It was designed and implemented by AppEx Networks Corporation.
Infineta Systems was a company that made WAN optimization products for high performance, latency-sensitive network applications. The company advertised that it allowed application data rate to exceed the nominal data rate of the link. Infineta Systems ceased operations by February 2013, a liquidator was appointed, and its products will no longer be manufactured, sold or distributed.
| | This computer science article is a stub. You can help Wikipedia by expanding it. |