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 study of deploying smooth‐ and responsive‐TCPs with different queue management schemes

While there exist extensive research works on congestion control and active queue management, or the joint dynamics of a congestion control strategy with the random early detection (RED) algorithm, little has been done on the interactions between different window adjustment strategies and different queue management schemes such as DropTail and RED. In this paper, we consider a spectrum of TCP‐friendly additive increase and multiplicative decrease (AIMD) parameters. At the one end of this spectrum, smooth‐TCP enhances smoothness for multimedia applications by reducing the window decrease ratio upon congestion, at the cost of the additive increase speed and the responsiveness to available bandwidth. At the other end, responsive‐TCP enhances the responsiveness by increasing the additive increase speed, at the cost of smoothness. We investigate the network dynamics with various combinations of AIMD parameters and queue management schemes, under different metrics. The investigation is conducted from the deployment (especially incremental deployment) point of view. We discussed the impact of the interactions on the goodput, fairness, end‐to‐end delay, and its implications to energy consumption on mobile hosts. Copyright © 2008 John Wiley & Sons, Ltd.     

基于反步滑模控制的TCP网络的主动队列管理

针对TCP网络的拥塞问题,考虑到网络本身存在参数不确定因素和非响应流的干扰,基于反步滑模控制提出了一种主动队列管理算法。在总的不确定的界已知而且不必很小的情况下,设计了一种反步滑模控制器来补偿系统不确定所带来的影响。仿真结果表明,该方法对TCP网络的复杂变化具有较好的鲁棒性和较快的系统响应。     

TCP网络的自适应非奇异终端滑模控制

针对TCP网络的拥塞控制问题,采用非奇异终端滑模控制理论提出了一种新的主动队列管理算法。采用非奇异终端滑模面以克服传统终端滑模控制的奇异问题,同时确保系统能在有限时间内收敛至平衡点。考虑到UDP流干扰的情况,用Lyapunov稳定性方法给出了一个自适应律来消除UDP流干扰对系统的影响。仿真结果表明,该算法可以使队列长度快速收敛到设定值,同时维持较小的队列振荡,优于传统的滑模控制。     

Using TCP rate control for queue input‐output rate matching

In explicit TCP rate control, the receiver's advertised window size in acknowledgment (ACK) packets can be modified by intermediate network elements to reflect network congestion conditions. The TCP receiver's advertised window (i.e. the receive buffer of a TCP connection) limits the maximum window and consequently the throughput that can be achieved by the sender. Appropriate reduction of the advertised window can control the number of packets allowed to be sent from a TCP source. This paper evaluates the performance of a TCP rate control scheme in which the receiver's advertised window size in ACK packets are modified in a network node in order to match the generated load to the assigned bandwidth in the node. Using simulation and performance metrics such as the packet loss rates and the cumulative number of TCP timeouts, we examine the service improvement provided by the TCP rate control scheme to the users. The modified advertised windows computed in the network elements and the link utilization are also examined. Copyright © 2002 John Wiley & Sons, Ltd.     

Fair end‐to‐end window‐based congestion control in time‐varying data communication networks

Communication networks are time varying and, hence, fair sharing of network resources among the users in such a dynamic environment is a challenging task. In this context, a time‐varying network model is designed, and the shortest user's route is found. In the designed network model, an end‐to‐end window‐based congestion control scheme is developed with the help of internal nodes or router, and the end user can get implicit feedback (throughput). This scheme is considered as fair if the allocation of resources among users minimizes overall congestion or backlog in the networks. Window update approach is based on a multi‐class fluid model and is updated dynamically by considering delays (communication, propagation, and queuing) and the backlog of packets in the user's routes. Convergence and stability of the window size are obtained using a Lyapunov function. A comparative study with other window‐based methods is also provided.     

Proxy‐RED: an AQM scheme for wireless local area networks

Wireless access points (APs) act as bridges between wired and wireless networks. Since the actually available bandwidth in wireless networks is much smaller than the bandwidth in wired networks, there is a disparity in channel capacity which makes the access point a significant network congestion point in the downstream direction. A current architectural trend in wireless local area networks (WLAN) is to move functionality from APs to a centralized gateway in order to reduce cost and improve features. In this paper, we study the use of RED, a well known active queue management (AQM) scheme, and explicit congestion notification (ECN) to handle bandwidth disparity between the wired and the wireless interface of an access point. Then, we propose the Proxy‐RED scheme, as a solution for reducing the AQM overhead from the access point. Simulations‐based performance analysis indicates that the proposed Proxy‐RED scheme improves the overall performance of a network. In particular, the Proxy‐RED scheme significantly reduces packet loss rate and improves goodput for a small buffer, and minimizes delay for a large buffer size. Copyright © 2006 John Wiley & Sons, Ltd.     

Relative loss rate differentiation: performance of short‐lived TCP flows

The relative differentiated service model provides assurances for the relative quality ordering between service classes, rather than for the actual service level in each class. In this paper, we describe a relative loss rate differentiation scheme where packet drop probabilities are determined according to an active queue management (AQM) mechanism based on random early detection (RED) in a first‐in first‐out (FIFO) queue, are weighted in inverse proportion to the price that the network operator assigns to each service class. Basically, we describe a scheme where relative loss rate differentiation is incorporated directly into AQM. Most TCP flows today, particularly Web flows, can be characterized as short‐lived flows. Using simulations with short‐lived TCP flows, we show that the scheme is very effective in ensuring relative loss rate differentiation between service classes during times of network congestion. Copyright © 2004 John Wiley & Sons, Ltd.     

Corruption‐aware adaptive increase and adaptive decrease algorithm for TCP error and congestion controls in wireless networks

The conventional TCP tends to suffer from performance degradation due to packet corruptions in the wireless lossy channels, since any corruption event is regarded as an indication of network congestion. This paper proposes a TCP error and congestion control scheme using corruption‐aware adaptive increase and adaptive decrease algorithm to improve TCP performance over wireless networks. In the proposed scheme, the available network bandwidth is estimated based on the amount of the received integral data as well as the received corrupted data. The slow start threshold is updated only when a lost but not corrupted segment is detected by sender, since the corrupted packets still arrive at the TCP receiver. In the proposed scheme, the duplicated ACKs are processed differently by sender depending on whether there are any lost but not corrupted segments at present. Simulation results show that the proposed scheme could significantly improve TCP throughput over the heterogeneous wired and wireless networks with a high bit error rate, compared with the existing TCP and its variants. Copyright © 2008 John Wiley & Sons, Ltd.     

REDu:一种新的识别并惩罚非适应流的主动式队列管理算法

本文提出一种新的主动式队列管理算法——热度算法(REDu).算法深入挖掘非适应流与适应流本质区别,利用CHOKe命中、RED丢弃等信息预选非适应流,通过热度升降机制计算一种新的部分流状态——热度,以此识别并惩罚非适应流.基于ns-2的仿真实验显示,与其他几种主动式队列管理算法相比,REDu具有更准确的识别并惩罚非适应流的能力,对适应流提供更好的保护,网络的鲁棒性也显著提高.     

Web browsing optimization over 2.5G and 3G: end‐to‐end mechanisms vs. usage of performance enhancing proxies

2.5 Generation (2.5G) and Third Generation (3G) cellular wireless networks allow mobile Internet access with bearers specifically designed for data communications. However, Internet protocols under‐utilize wireless wide area network (WWAN) link resources, mainly due to large round trip times (RTTs) and request‐‐reply protocol patterns. Web browsing is a popular service that suffers significant performance degradation over 2.5G and 3G. In this paper, we review and compare the two main approaches for improving web browsing performance over wireless links: (i) using adequate end‐to‐end parameters and mechanisms and (ii) interposing a performance enhancing proxy (PEP) between the wireless and wired parts. We conclude that PEPs are currently the only feasible way for significantly optimizing web browsing behavior over 2.5G and 3G. In addition, we evaluate the two main current commercial PEPs over live general packet radio service (GPRS) and universal mobile telecommunications system (UMTS) networks. The results show that PEPs can lead to near‐ideal web browsing performance in certain scenarios. Copyright © 2006 John Wiley & Sons, Ltd.     

Comparative analysis of cumulative distribution function for TCP congestion window size and triple‐duplicate period

Various version of Transmission Control Protocol (TCP) congestion window model are available in the literature. TCP congestion control is handled in two phases: (a) slow start and (b) congestion avoidance. This paper deals with congestion avoidance phase that is based on additive increase multiplicative decrease (AIMD) mechanism in which window size either increase by one or cuts to half of the previous window size. For distribution of window sizes, a model developed by Yan et al has been presented for two loss indications: (a) triple duplicate (TD) and (b) time‐outs (TOs) only while it does not deal with the window size limitation. It does not allow to cap over the TCP congestion window size if the buffer size is restricted to a fixed value though window size could not move beyond that. The unconstrained window size moves on in the presence of low loss probability. In this paper, we propose a model for the case where the window size is bounded by a maximum value that makes applicable to window size limitation as well. Further, the work has been extended to develop a new model that obtains a cumulative distribution function for TD periods (TDPs). The proposed model is validated on ns‐2, and we conclude that observed results for distribution function are very closed to our proposed model.     

Congestion control with dynamic threshold adaptation and cross‐layer response for TCP Vegas over IEEE 802.11 wireless networks

Wireless technologies provide mobile access and enable rapid andcost‐effective network deployment. But a wireless link is generally accompanied by high interference, transmission errors and a varying latency. The erratic packet losses usually lead to a curbing of the flow of segments on the TCP connection, and thus limit TCP performance. This paper presents a threshold control mechanism with cross‐layer response approach for improving TCP Vegas performance in IEEE 802.11 wireless networks. By making slight modifications to the legacy IEEE 802.11 MAC and TCP, the numerical results reveal that the proposed scheme provides a significant improvement in TCP performance under IEEE 802.11 wireless environments. Copyright © 2013 John Wiley & Sons, Ltd.     

Buffer management and scheduling schemes for TCP/IP over ATM‐GFR

Today ATM technology is facing challenges from integrated service IP, IP switching, gigabit IP router and gigabit ethernet. Although ATM is approved by ITU‐T as the standard technology in B‐ISDN, its survivability is still in question. Since ATM‐UBR (unspecified bit rate) provides no service guarantee and ATM‐ABR (available bit rate) is still unattainable for most users, many existing users have little or no incentives to migrate to ATM technology. The guaranteed frame rate (GFR) service is introduced to deal with this dilemma. The GFR can guarantee the minimum cell rate (MCR) with fair access to excess bandwidth. This paper studies various schemes to support the GFR. We have studied different discarding and scheduling schemes, and compared their throughput and fairness when TCP/IP traffic is carried. Through simulations, it is shown that only per‐VC queueing with weighted round robin (WRR) can guarantee minimum cell rate. Among all the schemes that have been explored, we recommend dynamic threshold–early packet discard (DT–EPD) integrated with MCR+ (a WRR variant) to support the GFR service. Copyright © 2001 John Wiley & Sons, Ltd.     

A study on a receiver‐based management scheme of access link resources for QoS‐controllable TCP connections

Although the bandwidth of access networks is rapidly increasing with the latest techniques such as DSL and FTTH, the access link bandwidth remains a bottleneck, especially when users activate multiple network applications simultaneously. Furthermore, since the throughput of a standard TCP connection is dependent on various network parameters, including round‐trip time and packet loss ratio, the access link bandwidth is not shared among the network applications according to the user's demands. In this thesis, we present a new management scheme of access link resources for effective utilization of the access link bandwidth and control of the TCP connection's throughput. Our proposed scheme adjusts the total amount of the receive socket buffer assigned to TCP connections to avoid congestion at the access network, and assigns it to each TCP connection according to characteristics in consideration of QoS. The control objectives of our scheme are (1) to protect short‐lived TCP connections from the bandwidth occupation by long‐lived TCP connections, and (2) to differentiate the throughput of the long‐lived TCP connections according to the upper‐layer application's demands. One of the results obtained from the simulation experiments is that our proposed scheme can reduce the delay of short‐lived document transfer perceived by the receiver host by up to about 90%, while a high utilization of access link bandwidth is maintained. Copyright © 2006 John Wiley & Sons, Ltd.