Fair end-to-end window-based congestion control

In this paper, we demonstrate the existence of fair end-to-end window-based congestion control protocols for packet-switched networks with first come-first served routers. Our definition of fairness generalizes proportional fairness and includes arbitrarily close approximations of max-min fairness. The protocols use only information that is available to end hosts and are designed to converge reasonably fast. Our study is based on a multiclass fluid model of the network. The convergence of the protocols is proved using a Lyapunov function. The technical challenge is in the practical implementation of the protocols     

Equation-based end-to-end single-rate multicast congestion control

Among the recently proposed single-rate multicast congestion control protocols is transmission control protocol-friendly multicast congestion control (TFMCC; Widmer and Handley 2001; Floyd et al. 2000; Widmer et al. IEEE Netw 15:28–37, 2001), which is an equation-based single-rate protocol that extends the mechanisms of the unicast TCP-friendly rate control (TFRC) protocol into the multicast domain. In TFMCC, each receiver estimates its throughput using an equation that estimates the steady-state throughput of a TCP source. The source then adjusts its sending rate according to the slowest receiver within the session (a.k.a., current-limiting receiver, CLR). TFMCC is a relatively simple, scalable, and TCP-friendly multicast congestion control protocol. However, TFMCC is hindering its throughput performance by adopting an equation derived from the unicast TFRC protocol. Further, TFMCC is slow to react to congestion conditions that usually result in a change of the CLR. This paper is motivated by these two observations and proposes an improved version of TFMCC, which we refer to as hybrid-TFMCC (or H-TFMCC for short). First, each receiver estimates its throughput using an equation that models the steady-state throughput of a multicast source controlled according to the additive increase multiplicative decrease (AIMD) approach. The second modification consists of adopting a hybrid sender/receiver-based rate control strategy, where the sending rate can be adjusted by the source or initiated by the current or a new CLR. The source monitors RTT variations on the CLR path, in order to rapidly adjust the sending rate to network conditions. Simulation results show that these modifications result in remarkable performance improvement with respect to throughput, time to react, and magnitude of oscillations. We also show that H-TFMCC remains TCP-friendly and achieves a higher fairness index than that achieved by TFMCC.     

Promoting the use of end-to-end congestion control in the Internet

This paper considers the potentially negative impacts of an increasing deployment of non-congestion-controlled best-effort traffic on the Internet. These negative impacts range from extreme unfairness against competing TCP traffic to the potential for congestion collapse. To promote the inclusion of end-to-end congestion control in the design of future protocols using best-effort traffic, we argue that router mechanisms are needed to identify and restrict the bandwidth of selected high-bandwidth best-effort flows in times of congestion. The paper discusses several general approaches for identifying those flows suitable for bandwidth regulation. These approaches are to identify a high-bandwidth flow in times of congestion as unresponsive, “not TCP-friendly”, or simply using disproportionate bandwidth. A flow that is not “TCP-friendly” is one whose long-term arrival rate exceeds that of any conformant TCP in the same circumstances. An unresponsive flow is one failing to reduce its offered load at a router in response to an increased packet drop rate, and a disproportionate-bandwidth flow is one that uses considerably more bandwidth than other flows in a time of congestion     

Designing a binary controller to enhance end-to-end congestion control

Active queue management (AQM) is proposed to enhance end-to-end congestion control through purposefully dropping packets in the intermediate nodes. In this letter, a novel packet dropping mechanism is developed through designing a binary controller applying the robust control theory. The new mechanism can simplify the manipulation on the AQM router so as to be helpful for implementing the high performance router. The numerical simulation results show that the binary controller can satisfy with the technical requirements for AQM     

信道公平分配的局部拥塞控制算法

提出基于信道公平分配的局部拥塞控制算法FCA(fair channel allocation),在缓解局部拥塞的同时增强信道分配的公平性。为减少获取邻居节点实时缓存信息的通信开销和提高以单一节点缓存是否溢出为检测模型的准确性,FCA采用以节点实时缓存长度预测为基础的邻居节点缓存总长度和分组平均传输延迟作为检测指标的拥塞检测模型。为避免使用独立拥塞通告消息增加信道负载,FCA采用在ACK控制帧中增加一个节点地址位携带拥塞信息。在去拥塞阶段,FCA采用基于实时缓存长度和队列优先权值的信道分配机制保证公平传输和防止部分节点因缓存增速过快导致溢出分组丢失。实验结果表明,FCA在碰撞次数、分组传递率、吞吐量和公平性等方面相比802.11、CODA和PCCP具有显著优势。     

On max-min fair congestion control for multicast ABR service in ATM

In the ATM Forum activities, considerable efforts have focused on the congestion control of point-to-point available bit rate (ABR) service. We present a novel approach that extends existing point-to-point (unicast) congestion control protocols to a point-to-multipoint (multicast) environment. In particular, we establish a unified framework to derive a multicast congestion control protocol for an ABR service from a given rate-based unicast protocol. We generalize a known necessary and sufficient condition on the max-min fairness of unicast rate allocation for a multicast service. Using this condition, we show that the resulting multicast protocol derived using our framework preserves the fairness characteristics of the underlying unicast protocol. The practical significance of our approach is illustrated by extending a standard congestion control mechanism for an ABR service to a multicast environment. The performance of the resulting multicast protocol is examined using benchmark network configurations suggested by the traffic management subworking group at the ATM Forum, and simulation results are presented to substantiate our claims     

计算机通信网络中基于速率的端对端拥塞控制

运用现代控制理论和方法，在多个信源节点单个信终端节点的端对端模型基础上，考虑到信终端节点的缓冲占有量的动态稳定，设计了一类可以比较好地调节ABR信源节点发送速率的控制器，使得终端节点的缓冲占有量可以逐渐稳定。     

Core‐stateless fair rate estimation fair queuing

Core‐stateless mechanisms, such as core‐stateless fair queuing (CSFQ), reduce the complexity of fair queuing, which usually need to maintain states, manage buffers, and perform flow scheduling on a per‐flow basis. However, they require executing label rewriting and dropping decision on a per‐packet basis, thus preventing them from being widely deployed. In this paper, we propose a novel architecture based on CSFQ without per‐packet labelling. Similarly, we distinguish between edge routers and core routers. Edge routers maintain the per‐flow state by employing a fair queuing mechanism to allocate each flow a fair bandwidth share locally and a token bucket mechanism to regulate those flows with feedback packets sent from egress edge routers. Core routers do not maintain per‐flow state; they use FIFO packet scheduling extended by a fare rate alarm mechanism by estimating the arrival rate and the number of flows using a matching–mismatching algorithm. The novel scheme is called core‐stateless fair rate estimation fair queuing (CSFREFQ). CSFREFQ is proven to be capable of achieving max–min fairness. Furthermore, we present and discuss simulations and experiments on the performance under different traffic scenarios. Copyright © 2005 John Wiley & Sons, Ltd.     

基于排队系统的最佳拥塞控制比例研究

对控制报文和网络拥塞间的平衡问题进行研究。通过一个单服务队列模型来描述拥塞控制策略,利用排队系统中的马尔可夫过程,提出一种两阈值的流量控制算法使其控制报文速率能满足最好的拥塞概率。通过分析发现排队系统中拥塞概率随缓冲区大小变化发生指数衰变,并定义该衰变指数为大偏差指数用来描述控制报文与拥塞概率间的比例。最后通过带宽共享模型,模拟并分析不同带宽情况下控制报文与拥塞概率间的最佳比例及其大偏差指数。     

VIRAL: coupling congestion control with fair video quality metric

Video streaming is often carried out by congestion controlled transport protocols to preserve network sustainability. However, the success of the growth of such non-live video flows is linked to the user quality of experience. Thus, one possible solution is to deploy complex quality of service systems inside the core network. Another possibility would be to keep the end-to-end principle while making aware transport protocols of video quality rather than throughput. The objective of this article is to investigate the latter by proposing a novel transport mechanism which targets video quality fairness among video flows. Our proposal, called VIRAL for virtual rate-quality curve, allows congestion controlled transport protocols to provide fairness in terms of both throughput and video quality. VIRAL is compliant with any rate-based congestion control mechanisms that enable a smooth sending rate for multimedia applications. Implemented inside TFRC a TCP-friendly protocol, we show that VIRAL enables both intra-fairness between video flows in terms of video quality and inter-fairness in terms of throughput between TCP and video flows.     

Efficient fair queuing using deficit round-robin

Fair queuing is a technique that allows each flow passing through a network device to have a fair share of network resources. Previous schemes for fair queuing that achieved nearly perfect fairness were expensive to implement; specifically, the work required to process a packet in these schemes was O(log(n)), where n is the number of active flows. This is expensive at high speeds. On the other hand, cheaper approximations of fair queuing reported in the literature exhibit unfair behavior. In this paper, we describe a new approximation of fair queuing, that we call deficit round-robin. Our scheme achieves nearly perfect fairness in terms of throughput, requires only O(1) work to process a packet, and is simple enough to implement in hardware. Deficit round-robin is also applicable to other scheduling problems where servicing cannot be broken up into smaller units (such as load balancing) and to distributed queues     

Hardware implementation of fair queuing algorithms for asynchronoustransfer mode networks

Providing quality-of-service guarantees in both cell- and packet-based networks requires the use of a scheduling algorithm in the switches and network interfaces. These algorithms need to be implemented in hardware in a high-speed switch. The authors present a number of approaches to implement scheduling algorithms in hardware. They begin by presenting a general methodology for the design of timestamp-based fair queuing algorithms that provide the same bounds on end-to-end delay and fairness as those of weighted fair queuing, yet have efficient hardware implementations. Based on this general methodology, the authors describe two specific algorithms, frame-based fair queuing and starting potential-based fair queuing, and discuss illustrative implementations in hardware. These algorithms may be used in both cell switches and packet switches with variable-size packets. A methodology for combining a traffic shaper with this class of fair queuing schedulers is also presented for use in network interface devices, such as an ATM segmentation and reassembly device     

Handover compensation scheme for weighted fair queuing in cellularpacket switched networks

The unbounded fairness problem of WFQ induced by handovers in cellular packet switched networks is investigated and a new strategy for guaranteed bounded fairness is proposed. The simulation results show that the proposed scheme maintains bounded fairness, while in WFQ an increase in unfairness from a handover is cumulative, allowing a new flow to monopolise a wireless link in proportion to the increase     

Detecting shared congestion of flows via end-to-end measurement

Current Internet congestion control protocols operate independently on a per-flow basis. Recent work has demonstrated that cooperative congestion control strategies between flows can improve performance for a variety of applications, ranging from aggregated TCP transmissions to multiple-sender multicast applications. However, in order for this cooperation to be effective, one must first identify the flows that are congested at the same set of resources. We present techniques based on loss or delay observations at end hosts to infer whether or not two flows experiencing congestion are congested at the same network resources. Our novel result is that such detection can be achieved for unicast flows, but the techniques can also be applied to multicast flows. We validate these techniques via queueing analysis, simulation and experimentation within the Internet. In addition, we demonstrate preliminary simulation results that show that the delay-based technique can determine whether two TCP flows are congested at the same set of resources. We also propose metrics that can be used as a measure of the amount of congestion sharing between two flows     

A radio‐aware worst‐case fair weighted fair queuing scheduler for WiMAX networks

This paper presents RA‐WF²Q+, a radio‐aware version of the well‐known WF²Q+ scheduler, tailored for the downlink traffic of a WiMAX system. With respect to WF²Q+, the proposed scheduler introduces the number of slots in the downlink frame as a parameter of the sharing decision process. This solution permits taking into account the information on the radio channel quality experimented by each user and, at the same time, exploiting the well‐known features of WF²Q+. The simulation study comparing the performance of the proposed scheduler with those of other well‐known scheduling algorithms highlights some interesting characteristics of RA‐WF²Q+. In particular, it permits achieving the highest overall system throughput while guaranteeing the requested quality of service for voice applications. Only when the radio resources become scarce with respect to the total offered traffic, RA‐WF²Q+ displays its opportunistic nature, favouring the users nearest to the base station. Copyright © 2012 John Wiley & Sons, Ltd.     

Simulation‐based results of weighted fair queuing algorithms for ATM networks

The behavior of the ideal General Processor Sharing (GPS) discipline and different per‐VC queuing algorithms approximating this ideal scheme, namely the Self Clocked Fair Queuing, the Packet by Packet Generalized Processor Sharing, and the Virtual Starting Time disciplines, are studied in this paper via simulation. We specifically consider a simple simulation configuration involving two Constant Bit rate (CBR) connections and several ON/OFF connections (bursty traffic). This simple simulation experiment allows us to point out three important features of the GPS and approximating disciplines. First, by adequately choosing the weight coefficients, these scheduling schemes can offer to CBR traffic almost Head of Line (HOL) priority over ON/OFF connections, to each of which, nevertheless, a minimum bandwidth is guaranteed. Second, GPS and per‐VC queuing disciplines, like the simple FIFO scheme, is very sensitive to burst scale congestion phenomena. Finally, simulation results seem to indicate that the scheduling disciplines considered perform traffic shaping on ON/OFF connections, which drastically reduces the burstiness of output traffic. This revised version was published online in June 2006 with corrections to the Cover Date.     

Soft QoS provisioning using the token bank fair queuing scheduling algorithm

Future-generation wireless packet networks will support multimedia applications with diverse QoS requirements. Much of the research on scheduling algorithms has been focused on hard QoS provisioning of integrated services. Although these algorithms give hard delay bounds, their stringent requirements sacrifice the potential statistical multiplexing performance and flexibility of the packet-switched network. Furthermore, the complexities of the algorithms often make them impractical for wireless networks. There is a need to develop a packet scheduling scheme for wireless packet-switched networks that provides soft QoS guarantees for heterogeneous traffic, and is also simple to implement and manage. This article proposes token bank fair queuing (TBFQ), a soft scheduling algorithm that possesses these qualities. This algorithm is work-conserving and has a complexity of O(1). We focus on packet scheduling on a reservation-based TDMA/TDD wireless channel to service integrated real-time traffic. The TBFQ scheduling mechanism integrates the policing and servicing functions, and keeps track of the usage of each connection. We address the impact of TBFQ on mean packet delay, violation probability, and bandwidth utilization. We also demonstrate that due to its soft provisioning capabilities, the TBFQ performs rather well even when traffic conditions deviate from the established contracts.     

iTCP: an intelligent TCP with neural network based end-to-end congestion control for ad-hoc multi-hop wireless mesh networks

Maintaining the performance of reliable transport protocols, such as transmission control protocol (TCP), over wireless mesh networks (WMNs) is a challenging problem due to the unique characteristics of data transmission over WMNs. The unique characteristics include multi-hop communication over lossy and non-deterministic wireless mediums, data transmission in the absence of a base station, similar traffic patterns over neighboring mesh nodes, etc. One of the reasons for the poor performance of conventional TCP variants over WMNs is that the congestion control mechanisms in conventional TCP variants do not explicitly account for these unique characteristics. To address this problem, this paper proposes a novel artificial intelligence based congestion control technique for reliable data transfer over WMNs. The synergy with artificial intelligence is established by exploiting a carefully designed neural network (NN) in the congestion control mechanism. We analyze the proposed NN based congestion control technique in detail and incorporate it into TCP to create a new variant that we name as intelligent TCP or iTCP. We evaluate the performance of iTCP using both ns-2 simulations and real testbed experiments. Our evaluation results demonstrate that our proposed congestion control technique exhibits a significant improvement in total network throughput and average energy consumption per transmitted bit compared to the congestion control techniques used in other TCP variants.     

Accumulation-based congestion control

This paper generalizes the TCP Vegas congestion avoidance mechanism and uses accumulation , buffered packets of a flow inside network routers, as a congestion measure based on which a family of congestion control schemes can be derived. We call this model Accumulation-based Congestion Control (ACC), which fits into the nonlinear optimization framework proposed by Kelly. The ACC model serves as a reference for packet-switching network implementations. We show that TCP Vegas is one possible scheme under this model. It is well known that Vegas suffers from round trip propagation delay estimation error and reverse path congestion. We therefore design a new Monaco scheme that solves these problems by employing an out-of-band, receiver-based accumulation estimator, with the support of two FIFO priority queues from the (congested) routers. Comparisons between these two schemes demonstrate that Monaco does not suffer from the problems mentioned above and achieves better performance than Vegas. We use ns-2 simulations and Linux implementation experiments to show that the static and dynamic performance of Monaco matches the theoretic results. One key issue regarding the ACC model in general, i.e., the scalability of bottleneck buffer requirement, and a solution using a virtual queueing algorithm are discussed and evaluated.     

Optimization-based congestion control for multicast communications

This article outlines an approach for multicast congestion control based on an economic model that has been successfully applied to unicast congestion control. In this model, congestion signals are interpreted as prices and congestion-controlled sessions as utility maximizing agents. A naive extension of the unicast model fails to achieve a reasonable balance between providing the incentives necessary to promote the use of multicast and ensuring that multicast sessions do not interact too aggressively with unicast sessions. We extend the model by introducing a rational definition of multicast utility. The revised model provides a basis for multicast congestion control protocols that provide incentives to use multicast but are necessarily unfair to unicast traffic. We show, however, that the degree of unfairness can be controlled by appropriately setting a design parameter with a limiting case of strict fairness