DRED: a random early detection algorithm for TCP/IP networks

It is now widely accepted that a RED [2] controlled queue certainly performs better than a drop‐tail queue. But an inherent weakness of RED is that its equilibrium queue length cannot be maintained at a preset value independent of the number of TCP active connections. In addition, RED's optimal parameter setting is largely correlated with the number of connections, the round‐trip time, the buffer space, etc. In light of these observations, we propose DRED, a novel algorithm which uses the basic ideas of feedback control to randomly discard packets with a load‐dependent probability when a buffer in a router gets congested. Over a wide range of load levels, DRED is able to stabilize a router queue occupancy at a level independent of the number of active TCP connections. The benefits of stabilized queues in a network are high resources utilization, predictable maximum delays, more certain buffer provisioning, and traffic‐load‐independent network performance in terms of traffic intensity and number of connections. Copyright © 2002 John Wiley & Sons, Ltd.     

A comparison of active queue management algorithms using the OPNETModeler

A number of active queue management algorithms for TCP/IP networks such as random early detection (RED), stabilized RED (SRED), BLUE, and dynamic RED (DRED) have been proposed in the past few years. This article presents a comparative study of these algorithms using simulations. The evaluation is done using the OPNET Modeler, which provides a convenient and easy-to-use platform for simulating large-scale networks. The performance metrics used in the study are queue size, packet drop probability, and packet loss rate. The study shows that, among the four algorithms, SIZED and DRED are more effective at stabilizing the queue size and controlling the packet loss rate while maintaining high link utilization. The benefits of stabilized queues in a network are high resource utilization, bounded delays, more certain buffer provisioning, and,traffic-load-independent network performance in terms of traffic intensity and number of TCP connections     

Buffer management schemes for supporting TCP in gigabit routerswith per-flow queueing

There has been much interest in using active queue management in routers in order to protect users from connections that are not very responsive to congestion notification. An Internet draft recommends schemes based on random early detection for achieving these goals, to the extent that it is possible, in a system without “per-flow” state. However, a “stateless” system with first-in/first-out (FIFO) queueing is very much handicapped in the degree to which flow isolation and fairness can be achieved. Starting with the observation that a “stateless” system is but one extreme in a spectrum of design choices and that per-flow queueing for a large number of flows is possible, we present active queue management mechanisms that are tailored to provide a high degree of isolation and fairness for TCP connections in a gigabit IP router using per-flow queueing. We show that IP flow state in a router can be bounded if the scheduling discipline used has finite memory, and we investigate the performance implications of different buffer management strategies in such a system. We show that merely using per-flow scheduling is not sufficient to achieve effective isolation and fairness, and it must be combined with appropriate buffer management strategies     

Understanding CHOKe: throughput and spatial characteristics

A recently proposed active queue management, CHOKe, is stateless, simple to implement, yet surprisingly effective in protecting TCP from UDP flows. We present an equilibrium model of TCP/CHOKe. We prove that, provided the number of TCP flows is large, the UDP bandwidth share peaks at (e+1)/sup -1/=0.269 when UDP input rate is slightly larger than link capacity, and drops to zero as UDP input rate tends to infinity. We clarify the spatial characteristics of the leaky buffer under CHOKe that produce this throughput behavior. Specifically, we prove that, as UDP input rate increases, even though the total number of UDP packets in the queue increases, their spatial distribution becomes more and more concentrated near the tail of the queue, and drops rapidly to zero toward the head of the queue. In stark contrast to a nonleaky FIFO buffer where UDP bandwidth shares would approach 1 as its input rate increases without bound, under CHOKe, UDP simultaneously maintains a large number of packets in the queue and receives a vanishingly small bandwidth share, the mechanism through which CHOKe protects TCP flows.     

几种主动式队列管理算法的比较研究

主动式队列管理(Active Queue Management,AQM)技术是IETF为了解决Internet拥塞控制问题而提出的一种路由器缓存管理技术.本文对几种主要AQM算法RED、BLUE、ARED和SRED的性能在基于ns-2仿真实验的基础上进行了比较研究.研究的性能包括队列长度、丢包概率、丢包率、连接数对吞吐量的影响及缓冲区大小对链路利用率的影响等.仿真结果表明BLUE、ARED和SRED在这几方面的性能都要优于RED算法.     

Fixed-point analysis of a network of routers with persistent TCP/UDP flows and class-based weighted fair queuing

Fixed-point models have already been successfully used to analytically study networks consisting of persistent TCP flows only, or mixed TCP/UDP flows with a single queue per link and differentiated buffer management for these two types of flows. In the current study, we propose a nested fixed-point analytical method to obtain the throughput of persistent TCP and UDP flows in a network of routers supporting class-based weighted fair queuing allowing the use of separate queues for each class. In particular, we study the case of two classes where one of the classes uses drop-tail queue management and is intended for only UDP traffic. The other class targeting TCP, but also allowing UDP traffic for the purpose of generality, is assumed to employ active queue management. The effectiveness of the proposed analytical method is validated in terms of accuracy using ns-3 simulations and the required computational effort.     

Towards a better-than-best-effort forwarding service for multimediaflows

A salient requirement of interactive multimedia applications is that they transmit data continuously at uniform rates with minimum possible end-to-end delay. The majority of these applications do not require hard and fast guarantees of network performance, but the current best-effort forwarding model of the Internet is frequently insufficient for realizing these requirements. Worse still, the requirement of uniform-rate transmission puts many multimedia applications at odds with current and proposed Internet network management practices that assume or require TCP-like reactions to packet loss. We are investigating router-based active queue management, specifically the use of queue occupancy thresholds to isolate TCP flows and to provide a better-than-best-effort forwarding service for flows in need of uniform-rate transmissions. Our current scheme, class-based thresholds (CBT), relies on a packet marking mechanism such as those proposed for realizing differentiated services on the Internet. CBT, when combined with existing active router queue management schemes such as random early detection (RED), provides a performance for TCP that approximates that achievable under a packet scheduling scheme and acceptable performance for multimedia flows. CBT is a simple and efficient mechanism with implementation complexity and run-time overhead comparable to that of RED     

动态权重RED网关

随着网络流量的增加,拥塞控制的重要性日益突出,尤其是链路拥塞控制。IETF推荐RED(Random Early Detection)作为下一代Internet路由器的拥塞控制算法。该文在分析了RED的优点和缺点后,在现在RED算法的基础上通过改进RED算法的平均队列长度计算,使用平均队列长度的均值,判断拥塞的发生和结束,动态调整权重,提高RED对拥塞发生和拥塞结束的反应速度,缩短拥塞时间,维持较低的实际队列长度。最后用仿真验证了该方法。     

一种参数自适应的主动队列管理算法-自适应BLUE

BLUE算法是一种典型的主动队列管理(Active Queue Management,AQM)算法,研究表明BLUE算法优于RED算法.BLUE算法使用丢包事件和链路空闲事件控制网络拥塞.但由于BLUE算法在参数设置方面存在不足,尤其是当TCP连接数突然剧烈变动时,容易导致队列溢出或空闲的频繁发生.该文引进参数自适应机制,提出了自适应BLUE算法,并借鉴了RED算法的早期拥塞检测机制.NS仿真实验表明该算法能有效保持队列长度的稳定,减少队列溢出或空闲现象的发生,在提高链路利用率的同时降低丢包率.     

A novel congestion control protocol with AQM support for IP-based networks

Multimedia services (Real-time and Non real-time) have different demands, including the need for high bandwidth and low delay, jitter and loss. TCP is a dominant protocol on the Internet. In order to have the best performance in TCP, the congestion window size must be set according to some parameters, since the TCP source is not aware of the window size. TCP emphasizes more on reliability than timeliness, so TCP is not suitable for real-time traffic. In this paper an active Queue management support TCP (QTCP) model is presented. Source rate is regulated based on the feedback which is received from intermediate routers. Furthermore, in order to satisfy the requirements of multimedia applications, a new Optimization Based active Queue management (OBQ) mechanism has been developed. OBQ calculates packet loss probabilities based on the queue length, packets priority and delay in routers and the results are sent to source, which can then regulate its sending rate. Simulation results indicate that the QTCP reduces packet loss and buffer size in intermediate nodes, improves network throughput and reduces delay.     

Dynamic buffer management using per‐queue thresholds

Shared buffer switches consist of a memory pool completely shared among output ports of a switch. Shared buffer switches achieve low packet loss performance as buffer space is allocated in a flexible manner. However, this type of buffered switches suffers from high packet losses when the input traffic is imbalanced and bursty. Heavily loaded output ports dominate the usage of shared memory and lightly loaded ports cannot have access to these buffers. To regulate the lengths of very active queues and avoid performance degradations, threshold‐based dynamic buffer management policy, decay function threshold, is proposed in this paper. Decay function threshold is a per‐queue threshold scheme that uses a tailored threshold for each output port queue. This scheme suggests that buffer space occupied by an output port decays as the queue size of this port increases and/or empty buffer space decreases. Results have shown that decay function threshold policy is as good as well‐known dynamic thresholds scheme, and more robust when multicast traffic is used. The main advantage of using this policy is that besides best‐effort traffic it provides support to quality of service (QoS) traffic by using an integrated buffer management and scheduling framework. Copyright © 2006 John Wiley & Sons, Ltd.     

数据流的活动队列管理算法:MBLUE

MBLUE(Modified BLUE)是一种面向数据流的活动队列管理算法.它不是使用平均队列长度指示缓冲区拥塞状态,而是使用数据报丢弃的频率和队列空闲程度来管理网络拥塞.探测瓶颈连接早期的拥塞信息,通过数据报的丢弃和标记避免拥塞.它只维护一个先进先出队列,以较少的数据流状态信息,在不同流之间公平的分配网络带宽.能够适应瞬时的猝发流,能合理控制非TCP数据流,又能够保持较短的平均队列长度,从而控制、减轻网络拥塞.通过TCP/IP网络的模拟,证实算法在公平的分配网络带宽和降低数据报的丢失率上具有较好的鲁棒性.     

A framework to determine the optimal weight parameter of RED in next‐generation Internet routers

Active buffer management can improve the performance of Transmission Control Protocol/Internet Protocol‐based networks. Random early detection (RED), an active queue management scheme, has been proposed by the Internet Engineering Task Force for next‐generation Internet routers. RED uses a number of parameters, such as buffer thresholds, a packet drop probability and a weight parameter. RED suffers from low throughput and large delay/jitter and induces instability in networks. Previous attempts to improve the performance of RED were based on modifying the thresholds and drop probabilities. In this paper, we show that an optimal value of the weight parameter can improve the performance of RED, and then develop a framework to determine the optimal value of the weight parameter. We show that the optimal weight parameter obtained from our framework improves the performance of RED. Copyright © 2008 John Wiley & Sons, Ltd.     

A cognitive accountability mechanism for penalizing misbehaving ECN‐based TCP stacks

The introduction of high‐bandwidth demanding services such as multimedia services has resulted in important changes on how services in the Internet are accessed and what quality‐of‐experience requirements (i.e. limited amount of packet loss, fairness between connections) are expected to ensure a smooth service delivery. In the current congestion control mechanisms, misbehaving Transmission Control Protocol (TCP) stacks can easily achieve an unfair advantage over the other connections by not responding to Explicit Congestion Notification (ECN) warnings, sent by the active queue management (AQM) system when congestion in the network is imminent. In this article, we present an accountability mechanism that holds connections accountable for their actions through the detection and penalization of misbehaving TCP stacks with the goal of restoring the fairness in the network. The mechanism is specifically targeted at deployment in multimedia access networks as these environments are most prone to fairness issues due to misbehaving TCP stacks (i.e. long‐lived connections and a moderate connection pool size). We argue that a cognitive approach is best suited to cope with the dynamicity of the environment and therefore present a cognitive detection algorithm that combines machine learning algorithms to classify connections into well‐behaving and misbehaving profiles. This is in turn used by a differentiated AQM mechanism that provides a different treatment for the well‐behaving and misbehaving profiles. The performance of the cognitive accountability mechanism has been characterized both in terms of the accuracy of the cognitive detection algorithm and the overall impact of the mechanism on network fairness. Copyright © 2012 John Wiley & Sons, Ltd.     

Two-way TCP traffic over rate controlled channels: effects andanalysis

We study the performance of bidirectional TCP/IP connections over a network that uses rate-based flow and congestion control. An example of such a network is an asynchronous transfer mode (ATM) network using the available bit rate (ABR) service. The sharing of a common buffer by TCP packets and acknowledgment (acks) has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. It has been the expectation that maintaining a smooth flow of data using rate-based flow control would mitigate, if not eliminate, the various forms of burstiness experienced with the TCP window flow control. However, we show that the problem of TCP ack compression appears even while operating over a rate-controlled channel. The degradation in throughput due to bidirectional traffic can be significant. For example, even in the simple case of symmetrical connections with adequate window sizes, the throughput of each connection is only 66.67% of that under one-way traffic. By analyzing the periodic bursty behavior of the source IP queue, we derive estimates for the maximum queue size and arrive at a simple predictor for the degraded throughput, for relatively general situations. We validate our analysis using simulation on an ATM network using the explicit rate option of the ABR service. The analysis predicts the behavior of the queue and the throughput degradation in simple configurations and in more general situations     

一种基于模糊逻辑的主动队列管理算法

主动队列管理 (ActiveQueueManagement,AQM)技术作为Internet拥塞控制的一种有效方法 ,对于提高In ternet的服务质量具有十分重要的作用 .本文根据TCP拥塞控制算法基于数据包丢失的窗口变化机制 ,设计了一种基于模糊逻辑的主动队列管理算法 .该算法依据路由器中队列长度的变化情况 ,根据一定的模糊自校正原则来调整数据包的丢弃概率 ,从而使路由器中的队列长度稳定在参考值附近 .仿真结果表明该算法不但十分有效 ,而且对不同的网络状况具有很好的适应能力 .     

Explicit TCP window adaptation in a shared memory architecture

A multiple (priority) queueing system allows a network node to manage the queueing of packets in such a way that higher priority packets will always be served first, low priority packets will be discarded when the queue is full, and for same‐priority packets any interference between them will be prevented. This paper describes a TCP window control scheme for a shared memory device that has buffer memory logically organized into multiple queues. To handle changing queue traffic loads, the shared memory device uses a dynamic buffer threshold mechanism to allocate buffer space to the queues. The TCP window control scheme allows the receiver's advertised window size in ACK packets to be modified at the network queue in order to maintain the queue size at a computed dynamic threshold. Copyright © 2003 John Wiley & Sons, Ltd.     

多个TCP连接的拥塞丢包模型

本文提出了有关TCP连接的拥塞丢包分析模型.网络瓶颈一般承载许多TCP连接,瓶颈处不可避免的拥塞和缓存溢出,是导致网上丢包的主要原因.网络瓶颈处的行为很大程度上左右了网络性能.本文的模型估计了存在大量持续TCP连接时,网络瓶颈的丢包概率和网络传输中断概率,给出了对实际网络的良好近似.这对于研究TCP对网络性能的影响,提出改善网络性能的新算法,以及分析(从长远来看)TCP还应做哪些改进,都是非常有用的.     

A comparative study of fuzzy versus “fixed” thresholdsfor robust queue management in cell-switching networks

High-performance cell-based communications networks have been conceived to carry asynchronous traffic sources and support a continuum of transport rates ranging from low bit-rate to high bit-rate traffic. When a number of bursty traffic sources add cells, the network is inevitably subject to congestion. Traditional approaches to congestion management include admission control algorithms, smoothing functions, and the use of finite-sized buffers with queue management techniques. Most queue management schemes, reported in the literature, utilize “fixed” thresholds to determine when to permit or refuse entry of cells into the buffer. The aim is to achieve a desired tradeoff between the number of cells carried through the network, propagation delays of the cells, and the number of discarded cells. While binary thresholds are excessively restrictive, the rationale underlying the use of a large number of priorities appears to be ad hoc, unnatural, and unclear. The paper introduces the notion of cell-blocking, wherein a fuzzy thresholding function, based on Zadeh's (1965) fuzzy set theory, is utilized to deliberately refuse entry to a fraction of incoming cells from other switches. The blocked cells must be rerouted by the sending switch to other switches and, in the process, they may incur delays. The fraction of blocked cells is a continuous function of the current buffer occupancy level unlike the abrupt. The fuzzy cell-blocking scheme is simulated on a computer. Fuzzy queue management adapts superbly to sharp changes in cell arrival rates and maximum burstiness of bursty traffic sources     

A buffer management algorithm for improving up/down transmission congestion protocol fairness in IEEE 802.11 wireless local area networks

The fair allocation of the resources is an important issue in wireless local area network (WLAN) because all wireless nodes compete for the same wireless radio channel. When uplink and downlink transmission congestion protocol (TCP) flows coexist in WLAN, the network service is biased toward the uplink TCP flows, and the downlink TCP flows tend to starve. In this article, we investigate the special up/down TCP unfairness problem and point out that the direct cause is the uplink acknowledgement (ACK) packets occupy most buffer space of access point. We thus propose a buffer management algorithm to ensure the fairness among uplink and downlink TCP flows. In order to limit the greedy behavior of ACK packets, the proposed algorithm adjusts the maximum size of buffer allocated for the ACK packets. Analysis and simulation results show that the proposed solution not only provides the fairness but also achieves 10–20% lower queue delay and higher network goodput than the other solutions. Copyright © 2012 John Wiley & Sons, Ltd.