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2001
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10 pages
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This paper introduces and analyzes a class of nonlinear congestion control algorithms called binomial algorithms, motivated in part by the needs of streaming audio and video applications for which a drastic reduction in transmission rate upon each congestion indication (or loss) is problematic. Binomial algorithms generalize TCP-style additive-increase by increasing inversely proportional to a power ¢ of the current window (for TCP, ¢ ¤ £ ¦ ¥ ) ; they generalize TCP-style multiplicative-decrease by decreasing proportional to a power § of the current window (for TCP, § £ ©
Workshop on High Performance Switching and Routing, Merging Optical and IP Technologie, 2002
Absfrucr-Current TCP implementations employ Additive Increase Multiplicative Decrease (AIMD) as the congestion control mechanism. Recently, a new set of schemes called Binomial Congestion Control Schemes (BCCS) were proposed and a section of these schemes is TCP compliant. In this paper we evaluate the performance of these TCP compliant binomial schemes and show through simulations that AIMD performs better than the other BCCS policies in a wide range networking environments. Specifically, we study the performance of these schemes with respect to throughput, fairness, losses, timeouts and self-similarity. We show that the superior performance of AIMD can be attributed to its more conservative attitude in the presence of losses when it reduces its transmission rate much faster than the other schemes. This results in smaller congestion periods thereby reducing the losses and timeouts which in turn increases the throughput and decreases the degree of self-similarity of the traffic. We also evaluated the performance of TCP Compatible BCCS when they compete with TCP flows. It was found that with sufficiently large number of flows, BCCS competes fairly with TCP. However, with a smaller number of flows in the network TCP flows get smaller share of the bottleneck and disproportionately higher losses and timeouts.
The increasing diversity of Internet application requirements has spurred recent interest in transport protocols with flexible transmission controls. In window-based congestion control schemes, increase rules determine how to probe available bandwidth, whereas decrease rules determine how to back off when losses due to congestion are detected. The control rules are parameterized so as to ensure that the resulting protocol is TCP-friendly in terms of the relationship between throughput and loss rate. This paper presents a comprehensive study of a new spectrum of window-based congestion controls, which are TCP-friendly as well as TCP-compatible under RED. Our controls utilize history information in their control rules. By doing so, they improve the transient behavior, compared to recently proposed slowly responsive congestion controls such as general additive-increase and multiplicative-decrease (AIMD) and binomial controls. Our controls can achieve better tradeoffs among smoothness, aggressiveness, and responsiveness, and they can achieve faster convergence. We demonstrate analytically and through extensive ns simulations the steady-state and transient behavior of several instances of this new spectrum.
2003
Abstract The increasing diversity of Internet application requirements has spurred recent interest in transport protocols with flexible transmission controls. In window-based congestion control schemes, increase rules determine how to probe available bandwidth, whereas decrease rules determine how to back off when losses due to congestion are detected. The control rules are parameterized so as to ensure that the resulting protocol is TCP-friendly in terms of the relationship between throughput and loss rate.
IEEE/ACM Transactions on Networking, 2005
We consider a modification of TCP congestion control in which the congestion window is adapted to explicit bottleneck rate feedback; we call this RATCP (Rate Adaptive TCP). Our goal in this paper is to study and compare the performance of RATCP and TCP in various network scenarios with a view to understanding the possibilities and limits of providing better feedback to TCP than just implicit feedback via packet loss. To understand the dynamics of rate feedback and window control, we develop and analyze a model for a long-lived RATCP (and TCP) session that gets a time-varying rate on a bottleneck link. We also conduct experiments on a Linux based test-bed to study issues such as fairness, random losses, and randomly arriving short file transfers. We find that the analysis matches well with the results from the test-bed. For large file transfers, under low background load, ideal fair rate feedback improves the performance of TCP by 15%-20%. For small randomly arriving file transfers, though RATCP performs only slightly better than TCP it reduces losses and variability of throughputs across sessions. RATCP distinguishes between congestion and corruption losses, and ensures fairness for sessions with different round trip times sharing the bottleneck link. We believe that rate feedback mechanisms can be implemented using distributed flow control and recently proposed REM in which case, ECN bit itself can be used to provide the rate feedback.
Computer Communication Review, 2001
The recently developed notion of TCP-compatibility has led to a number of proposals for alternative congestion control algorithms whose long-term throughput as a function of a steady-state loss rate is similar to that of TCP. Motivated by the needs of some streaming and multicast applications, these algorithms seem poised to take the current TCP-dominated Internet to an Internet where many congestion control algorithms co-exist. An important characteristic of these alternative algorithms is that they are slowly-responsive, refraining from reacting as drastically as TCP to a single packet loss.
International Journal of Communication Networks and Distributed Systems, 2016
In order to curtail the escalating packet loss rates caused by an exponential increase in network traffic, active queue management techniques such as Random Early Detection (RED) have come into picture. Flow Random Early Drop (FRED) keeps state based on instantaneous queue occupancy of a given flow. FRED protects fragile flows by deterministically accepting flows from low bandwidth connections and fixes several shortcomings of RED by computing queue length during both arrival and departure of the packet. Stochastic Fair Queuing (SFQ) ensures fair access to network resources and prevents a busty flow from consuming more than its fair share. In case of (Random Exponential Marking) REM, the key idea is to decouple congestion measure from performance measure (loss, queue length or delay). Stabilized RED (SRED) is another approach of detecting nonresponsive flows. In this paper, we have shown a comparative analysis of throughput, delay and queue length for the various congestion control algorithms RED, SFQ and REM. We also included the comparative analysis of loss rate having different bandwidth for these algorithms.
Los Alamos Unclassified …, 2001
AbstractThe TCP congestion-control mechanism is an algorithm designed to probe the available bandwidth of the network path that TCP packets traverse. However, it is well-known that the TCP congestion-control mechanism does not perform well on networks with a large ...
The demand for fast transfer of large volumes of data, and the deployment of the network infrastructures is ever increasing. However, the dominant transport protocol of today, TCP, does not meet this demand because it favors reliability over timeliness and fails to fully utilize the network capacity due to limitations of its conservative congestion control algorithm. The slow response of TCP in fast long distance networks leaves sizeable unused bandwidth in such networks. A large variety of TCP variants have been proposed to improve the connection's throughput by adopting more aggressive congestion control algorithms. Some of the flavors of TCP congestion control are loss-based, high-speed TCP congestion control algorithms that uses packet losses as an indication of congestion; delay-based TCP congestion control that emphasizes packet delay rather than packet loss as a signal to determine the rate at which to send packets. Some efforts combine the features of loss-based and delay-based algorithms to achieve fair bandwidth allocation and fairness among flows. A comparative analysis between different flavors of TCP congestion control namely Standard TCP congestion control (TCP Reno), loss-based TCP congestion control (HighSpeed TCP, Scalable TCP, CUBIC TCP), delay-based TCP congestion control (TCP Vegas) and mixed loss-delay based TCP congestion control (Compound TCP) is presented here in terns of congestion window verses elapsed time after the connection is established.
Computer Networks, 2010
Our study is motivated by the need to enable quality of service (QoS), congestion control and fair rate allocation for all end applications. We propose a new approach to address these needs which is different from the current practice whereby end applications pursue their own rate control using TCP. Our approach comprises a network rate management protocol (RMP) that controls the rate of all flows (at an aggregate level based on routes) subject to QoS requirements. The RMP control also facilitates a new TCP sliding-window congestion control based on the fair target rates computed by the RMP. Each non-TCP aggregate flow is policed by its respective edge router and each TCP flow adapts its window size as to achieve the RMP suggested fair target rate. The stability analysis of the new TCP congestion control is performed in a linearly scalable framework, which is less restrictive than a fluid model. We show that our proposed control is linearly scalable and establish its global asymptotic stability under arbitrary and variable information time lags, aka totally asynchronous conditions. The stability and the vitality of our control is verified by two means. One is a simulation of a network comprising 74 core links and up to 768 flows, each using its own access link. The simulation is also used to compare our control with the congestion control algorithms used in Fast, Vegas and Reno TCPs. The second verification means is an actual implementation of the control in the Linux kernel and its experimentation in a WAN testbed network comprising six routers and long haul links running UDP flows as well as CUBIC, N-RENO and C-TCP flows. Our experiments demonstrate that our approach can guarantee fair rates for all flows and QoS to premium flows.
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