A Lightweight Fairness-Driven AQM for Regulating Bandwidth Utilization in Best-Effort Routers

The end-to-end congestion control mechanism of transmission control protocol (TCP) is critical to the robustness and fairness of the best-effort Internet. Since it is no longer practical to rely on end-systems to cooperatively deploy congestion control mechanisms, the network itself must now participate in regulating its own resource utilization. To that end, fairness-driven active queue management (AQM) is promising in sharing the scarce bandwidth among competing flows in a fair manner. However, most of the existing fairness-driven AQM schemes cannot provide efficient and fair bandwidth allocation while being scalable. This paper presents a novel fairness-driven AQM scheme, called CHORD (CHOKe with recent drop history) that seeks to maximize fair bandwidth sharing among aggregate flows while retaining the scalability in terms of the minimum possible state space and per-packet processing costs. Fairness is enforced by identifying and restricting high-bandwidth unresponsive flows at the time of congestion with a lightweight control function. The identification mechanism consists of a fixed-size cache to capture the history of recent drops with a state space equal to the size of the cache. The restriction mechanism is stateless with two matching trial phases and an adaptive drawing factor to take a strong punitive measure against the identified high-bandwidth unresponsive flows in proportion to the average buffer occupancy. Comprehensive performance evaluation indicates that among other well-known AQM schemes of comparable complexities, CHORD provides enhanced TCP goodput and intra-protocol fairness and is well-suited for fair bandwidth allocation to aggregate traffic across a wide range of packet and buffer sizes at a bottleneck router.     

CSFQ算法分析与改进

核心无状态公平队列调度（CSVQ）算法提供了如同有状态网那样好的公平带宽分配，但它的丢包算法不适用于TCP流。针对TCP流的特点，对CSFQ算法进行如下改进：将缓存队列长度与丢包概率关联起来，用一种类似于RED（random early drop）缓存管理方法解决了缓存频繁溢出导致的一些问题；对TCP流的丢包率进行修正，使用多余带宽来转发TCP包，解决TCP流与UDP流的带宽分配公平性。仿真试验表明，新算法NEW-CSFQ更好地提供数据流公平的频宽共享，对突发流响应较原算法有所提高，且算法复杂度简单，容易在高速核心路由器上实现。     

Improving fairness among TCP flows by stateless buffer control with early drop maximum

Transmission control protocol (TCP) has been recognized as the most important transport-layer protocol for the Internet. It is distinguished by its reliable transmission, flow control, and congestion control. However, the issue of fair bandwidth-sharing among competing flows was not properly addressed in TCP. As web-based applications and interactive applications grow more popular, the number of short-lived flows conveyed on the Internet continues to rise. With conventional TCP, short-lived flows will be unable to obtain a fair share of available bandwidth. As a result, short-lived flows will suffer from longer delays and a lower service rate. It is essential for the Internet to come up with an effective solution to this problem in order to accommodate the new traffic patterns.With a more equitable sharing of bottleneck bandwidth as its goal, a stateless queue management scheme featuring early drop maximum (EDM) is developed and presented in this article. The fundamental idea is to drop packets from those flows having more than an equal share of bandwidth. The congestion window size of a TCP sender is carried in the options field on each packet. The proposed scheme will be exercised on routers and make its decision on packet dropping according to the congestion windows. In case of link congestion, the queued packet with the largest congestion window will be dropped from the queue. This will lower the sending rate of its sender and release part of the occupied bandwidth for the use of other competing flows. By so doing, the entire system will approach an equilibrium point with a rapid and fair distribution of bandwidth. As a stateless approach, the proposed scheme inherits numerous advantages in implementation and scalability.Extensive simulations were conducted to verify the feasibility and the effectiveness of the proposed approach. As revealed in the simulation results, the proposed scheme outperforms existing stateless techniques, including Drop-Tail and Random Early Drop, in many respects, such as a fairer sharing of available bandwidth and a shorter response time for short-lived flows.     

基于ECN的单播流与多播流间带宽共享算法

An ECN-based implementing bandwidth-sharing algorithm for unicast and multicast flows is presented.The algorithm uses a bandwidth allocation strategy to give an incentive to multicast flows in bandwidth allocation according to algorithm of the number of receivers, and to assure the unicast flows get their bandwidth shares fairly.Provided best-effort networks, an ECN-based congestion control algorithm is used to implement differentiated service in bandwidth allocation between unicast flows and multicast flows. In implementation, we solve the problems such asreceiver‘s number estimation, the RTT estimation and compromise between convergence and stability.The simulation results show that the algorithm can implement bandwidth sharing for TCP flows and multicast flows. Atthe same time, the algorithm not only allocates more bandwidth to multicast flows, but promises TCP flows to get their fair bandwidth share.     

高吞吐量的核心无状态公平队列算法

提出了一种优化的核心无状态公平队列凋度算法(xCSFQ),在CSFQ的基础上,根据缓冲区占用率和数据流到达速率决定丢包概率,缓存管理上采用基于CHOKC原理的机制进行缓存管理,解决了CSFQ链路利用率低的问题,提高了带宽在UDP流和TCP流之间分配的公平性,最后对算法进行了仿真分析。     

一种核心无状态保存的自适应成比例公平带宽分配机制

提出了一种核心无状态的自适应比例公平带宽分配机制CSPAFA（core stateless proporitonal adaptive fair allocation)，在边界路由器完成基于每个流的状态处理，将所有流分成标记流和非标记流两种业务类型，采用DPS（dynamic packet state)技术将有关信息编码进IP分组头，在核心将输出链路带宽分成两部分，核心根据当前的网络负荷对标记流按服务规格成比例的分配输出链路带宽，对未标记流公平分配带宽，并且能自适应地调整两类业务的带宽共享比例，最后，给出了在NS网络仿真环境下的仿真实验结果。     

A novel high speed transport protocol based on explicit virtual load feedback

Since traditional TCP congestion control is ill-suited for high speed networks, designing a high speed replacement for TCP has become a challenge. From the simulations of some existing high speed protocols, we observe that these high speed protocols make the round-trip time bias problem and the multiple-bottleneck bias problem more serious than for standard TCP. To address these problems, we apply the population ecology theory to design a novel congestion control algorithm. By analogy, we treat the network flows as the species in nature, the sending rates of the flows as the population number, and the bottleneck bandwidth as the food resources. Then we extend the construction method of population ecology models to design a control model, and implement the corresponding end-to-end protocol with virtual load factor feedback, which is called Explicit Virtual Load Feedback TCP (EVLF-TCP). The virtual load factor is computed based on the information of the bandwidth, the aggregate incoming traffic and the queue length in routers, and then senders adjust the sending rate based on the virtual load factor. Theoretical analysis and simulation results validate that EVLF-TCP achieves high utilization, fair bandwidth allocation independent of round-trip time, and near-zero packet drops. These characteristics are desirable for high speed networks.     

Anti-windup compensator for active queue management in TCP networks

In this paper, we apply a dynamic anti-windup scheme for improving the performance of a conventional proportional–integral (PI) controller for active queue management (AQM) supporting TCP flows. When a PI controller is used for AQM, the windup phenomenon of the integral action can cause performance degradation because the packet drop probability is limited between 0 and 1. Therefore we suggest a TCP/AQM model with a saturating actuator and apply a dynamic anti-windup method for improving the performance of the conventional PI AQM scheme. The proposed scheme not only provides graceful performance degradation, but also guarantees the stability of the overall system with the linearized TCP model. We verify the performance of the proposed scheme through ns-2 simulations. The simulation results show that our scheme outperforms the conventional PI controller when the traffic load is not stationary, which is always the case in real network environment.     

一种短流优先的公平带宽分配机制

提出一种短流优先的公平带宽分配机制FPIP(fair PIP).通过区别处理短流和长流的报文,FPIP能够将带宽优先分配给短流,然后将剩余的带宽在长流之间公平分配.此外,FPIP采用主动队列管理机制AQM(active queue management)检测拥塞并控制队列长度.仿真结果表明,FPIP在保证公平性、控制队列长度、减小Web流的响应时间等方面具有良好的性能.     

全局主动带宽调整算法

近年来，UDP等非响应流抢占TCP等响应流的现象越来越严重。提出全局主动带宽调整算法（GABA），通过网络中核心路由器与边缘路由器之间的协作，在核心路由器获取非响应流的信息，在边缘路由器阻止非响应流的大量进入。通过NS2仿真表明，GABA可以有效提高网络利用率和带宽分配的公平性。     

Can high-speed networks survive with DropTail queues management?

It has been observed that TCP connections that go through multiple congested links (MCL) have a smaller transmission rate than the other connections. Such TCP behavior is a result of two components (i) the cumulative packet losses that a flow experiences at each router along its path; (ii) the longer round trip times (RTTs) suffered by such flows due to non-negligible queueing delays at congested routers. This double “bias” against connections with MCLs has been shown to approximate the so-called minimum potential delay fairness principle in the current Internet. Despite the recent proliferation of new congestion control proposals for TCP in high-speed networks, it is still unclear what kind of fairness principle could be achieved with such newly proposed congestion control protocols in high-speed networks with large-delays. Studies already show that some high-speed TCP variants may cause surprisingly severe RTT unfairness in high-speed networks with DropTail routers.This paper studies the problem of unfairness in high-speed networks with some well-known high-speed TCP variants in presence of multiple congested links and highlights the severity of such unfairness when DropTail queue management is adopted.Through a simple synchronized loss model analysis, we show how synchronized losses with DropTail in high-speed networks could lead to severe RTT unfairness and drop probability (DP) unfairness; while random marking AQM schemes, which break the packet loss synchrony mitigate such unfairness dramatically by ensuring that the packet loss probability of a flow is the sum of the loss probabilities on the congested routers it crosses.Extensive simulations are carried out and the results support our findings.     

Performance evaluation of bandwidth allocation methods in a geostationary satellite channel in the presence of internet traffic

Resource allocation represents an important issue for the next generation TCP/IP Quality of Service-based satellite networks. Many schemes, proposed in the recent literature, consider Internet traffic as the superposition of traffic sources without distinguishing between User Datagram Protocol (UDP) and Transmission Control Protocol (TCP) flows, even if UDP and TCP imply very different traffic characteristics. The basic idea of this work is that a resource allocation algorithm which is conscious of the difference may be more efficient because it can make use of the different behaviour of TCP and UDP in the presence of network congestion. Actually TCP reduces the source flow rate and, as a consequence, also the bandwidth occupancy when there is network congestion. The use of this feature within the bandwidth allocation scheme allows reducing the bandwidth waste due to over provisioning and using the residual bandwidth for other sources. The advantage is particularly clear over satellite channels where fading often affects the communication: having some residual bandwidth available for stations which have experienced fading can improve the satellite network performance.This paper presents a detailed performance evaluation of a bandwidth allocation scheme, called E-CAP-ABASC and studied for the satellite environment. The bandwidth is assigned to the earth stations that compose the network by a master station on the basis of a cost function whose main part is represented by a closed-form of the packet loss probabilities for the TCP and UDP traffic. The use of two different packet loss probability models for TCP and UDP allows exploiting the different features of the two traffic types, so improving the overall performance either in terms of packet loss or, on the other hand, in terms of the traffic admitted.The performance evaluation is carried out by varying the link degradation due to fading, the traffic load, and the flow balance between UDP and TCP. The results show a good performance of E-CAP-ABASC, compared with two other schemes. Advantages and drawbacks are discussed.     

PRED:一种配合队列调度的RED算法

论文讨论了队列管理和队列调度的关系,提出了一种新的基于势的RED机制:PRED,以增强SPFQ调度机制的公平性,来达到整个报文处理系统的性能改善。通过严格的仿真验证,同其他的队列管理机制相比,PRED同SPFQ的结合能够最大限度地利用系统资源,提高服务的公平性。     

一种支持公平带宽分配的主动队列管理方案的研究

现在广泛使用的主动队列管理(AQM)是目前队列管理的主流技术,但是它不能很好地区分来自响应流(如TCP会话)和非响应流(如UDP)的包。当拥塞出现的时候,将导致非响应流压制响应流并占有较多的带宽,而响应流却无法获得公平带宽。本文提出了一种闭环拥塞控制(CLCC)方案,大大改善了带宽分配的公平性。     

基于高层模糊变结构控制的主动队列管理算法

从控制论的角度出发,主动队列管理算法将IP层纳入到拥塞控制机制.该文根据TCP拥塞控制机制的线性化模型,利用线性化模型时延因子的多项式表达式,提出了基于高层模糊变结构控制主动队列管理算法,简化了控制器的设计.NS仿真结果显示本算法提高了拥塞控制的性能和鲁棒性,有效地补偿了时延因子对于控制性能的影响.     

确保服务中一种基于动态阈值的数据包标记算法

确保转发过程中带宽享用的公平性问题一直是区分服务网络研究的热点，影响这种公平性的因素包括回路响应时间RTT、数据包大小、目标速率及聚流中包含的单流数量等．确保服务的实现依赖于在边界路由器执行的数据包标记策略和在核心路由器执行的队列管理策略，基于动态阈值的数据包标记策略DTBM的目的就是处理异质的TCP流之间带宽享用的公平性问题。DTBM通过测量局部吞吐量来调整标记算法中的阈值，以改变不同颜色的标记概率从而达到公平带宽享用的目的。DTBM的主要优点在于其实现简单、对参数不是很敏感并且对端结点主机是“透明”的，仿真实验表明，和其他几种标记算法相比，DTBM能有效地消除上述因素的影响，具有更好的公平性。     

基于限速的Ad hoc网络混合流拥塞控制研究

主动队列管理(AQM)的基础是TCP反馈机制,所以AQM在处理UDP与TCP混合流时无法控制UDP流量,导致非视频流影响视频UDP的传输质量。根据TCP拥塞窗口特性和混合流排队机制,推导了Ad hoc网络TCP/UDP的AQM模型,据此提出了一种基于UDP限速的PI主动队列管理算法。限速算法根据实际流速与设定流速之差,标记非视频UDP分组优先级并按从低到高的顺序丢弃分组。NS仿真表明,与PI控制相比,新算法实现了非视频UDP的流量控制,提高了视频传输质量0.98dB的峰值信噪比。     

A novel self-tuning feedback controller for active queue management supporting TCP flows

Wireless access points act as bridges between wireless and wired networks. Since the actually available bandwidth in wireless networks is much smaller than that in wired networks, there is a bandwidth disparity in channel capacity which makes the access point a significant network congestion point. The recently proposed active queue management (AQM) is an effective method used in wired network and wired-wireless network routers for congestion control, and to achieve a tradeoff between channel utilization and delay. The de facto standard, the random early detection (RED) AQM scheme, and most of its variants use average queue length as a congestion indicator to trigger packet dropping. In this paper, we propose a Novel autonomous Proportional and Differential RED algorithm, called NPD-RED, as an extension of RED. NPD-RED is based on a self-tuning feedback proportional and differential controller, which not only considers the instantaneous queue length at the current time point, but also takes into consideration the ratio of the current differential error signal to the buffer size. Furthermore, we give theoretical analysis of the system stability and give guidelines for the selection of feedback gains for the TCP/RED system to stabilize the instantaneous queue length at a desirable level. Extensive simulations have been conducted with ns2. The simulation results have demonstrated that the proposed NPD-RED algorithm outperforms the existing AQM schemes in terms of average queue length, average throughput, and stability.     

Analysis and design of controllers for AQM routers supporting TCPflows

In active queue management (AQM), core routers signal transmission control protocol (TCP) sources with the objective of managing queue utilization and delay. It is essentially a feedback control problem. Based on a recently developed dynamic model of TCP congestion-avoidance mode, this paper does three things: 1) it relates key network parameters such as the number of TCP sessions, link capacity and round-trip time to the underlying feedback control problem; 2) it analyzes the present de facto AQM standard: random early detection (RED) and determines that REDs queue-averaging is not beneficial; and 3) it recommends alternative AQM schemes which amount to classical proportional and proportional-integral control. We illustrate our results using ns simulations and demonstrate the practical impact of proportional-integral control on managing queue utilization and delay     

Delivering relative differentiated services in future high-speed networks using hierarchical dynamic deficit round robin

With the increasing deployment of real-time audio/video services over the Internet, provision of quality of service (QoS) has attracted much attention. When the line rate of future networks upgrades to multi-terabits per second, if routers/switches intend to deliver differentiated services through packet scheduling, the reduction of computational overhead and elimination of bottleneck resulting from memory latency will both become important factors. In addition, the decrease of average queueing delay and provision of small delays for short packets are two further critical factors influencing the delivery of better QoS for real-time applications. The advanced waiting time priority (AWTP) is a timestamp-based packet scheduler which is enhanced from the well-known WTP. Although AWTP considers the effect of packet size, the latency resulting from timestamp access and a great quantity of computational overhead may result in bottlenecks for AWTP being deployed over high-speed links. Many existing schedulers have the same problems. We propose a multi-level hierarchical dynamic deficit round-robin (MLHDDRR) scheduling scheme which is enhanced from the existing dynamic deficit round-robin scheduler. The new scheme can resolve these issues and efficiently provide relative differentiated services under a variety of load conditions. Besides, MLHDDRR can also protect the highest priority traffic from significant performance degradation due to bursts of low-priority traffic. We compare the performance of AWTP with the proposed scheme. Extensive simulation results and complexity analysis are presented to illustrate the effectiveness and efficiency of MLHDDRR.