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.
In computer networking, the transport layer is a conceptual division of methods in the layered architecture of protocols in the network stack in the Internet protocol suite and the OSI model. The protocols of this layer provide end-to-end communication services for applications. It provides services such as connection-oriented communication, reliability, flow control, and multiplexing.
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.
In computer networking, the Datagram Congestion Control Protocol (DCCP) is a message-oriented transport layer protocol. DCCP implements reliable connection setup, teardown, Explicit Congestion Notification (ECN), congestion control, and feature negotiation. The IETF published DCCP as RFC 4340, a proposed standard, in March 2006. RFC 4336 provides an introduction.
Network performance refers to measures of service quality of a network as seen by the customer.
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 Vegas is a TCP congestion avoidance algorithm that emphasizes packet delay, rather than packet loss, as a signal to help determine the rate at which to send packets. It was developed at the University of Arizona by Lawrence Brakmo and Larry L. Peterson and introduced in 1994.
TCP Westwood (TCPW) is a sender-side-only modification to TCP New Reno that is intended to better handle large bandwidth-delay product paths, with potential packet loss due to transmission or other errors, and with dynamic load.
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, The Transmission Control Protocol (TCP) detects packet loss and performs retransmissions to ensure reliable messaging. Packet loss in a TCP connection is also used to avoid congestion and thus produces an intentionally reduced throughput for the connection.
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.
Bandwidth management is the process of measuring and controlling the communications on a network link, to avoid filling the link to capacity or overfilling the link, which would result in network congestion and poor performance of the network. Bandwidth is described by bit rate and measured in units of bits per second (bit/s) or bytes per second (B/s).
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.
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).
Type of Transmission Control Protocol which is designed to provide much higher throughput and scalability.
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.
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.