REFWA: An Efficient and Fair Congestion Control Scheme for LEO Satellite Networks

This paper examines some issues that affect the efficiency and fairness of the Transmission Control Protocol (TCP), the backbone of Internet protocol communication, in multi-hops satellite network systems. It proposes a scheme that allows satellite systems to automatically adapt to any change in the number of active TCP flows due to handover occurrence, the free buffer size, and the bandwidth–delay product of the network. The proposed scheme has two major design goals: increasing the system efficiency, and improving its fairness. The system efficiency is controlled by matching the aggregate traffic rate to the sum of the link capacity and total buffer size. On the other hand, the system min-max fairness is achieved by allocating bandwidth among individual flows in proportion with their RTTs. The proposed scheme is dubbed Recursive, Explicit, and Fair Window Adjustment (REFWA). Simulation results elucidate that the REFWA scheme substantially improves the system fairness, reduces the number of packet drops, and makes better utilization of the bottleneck link. The results demonstrate also that the proposed scheme works properly in more complicated environments where connections traverse multiple bottlenecks and the available bandwidth may change over data transmission time.     

Enhancing Fairness for Short-Lived TCP Flows in 802.11b WLANs

The problem of providing throughput fairness in a wired-cum-wireless network where the wireless portion is an 802.11 wireless local area network (WLAN) is addressed. Due to the distributed nature of the primary 802.11 media access control protocol and the unpredictability of the wireless channel, quality of service guarantees in general and fairness in particular are hard to achieve in WLANs. This fact seriously compromises the interaction between 802.11-based networks and well-established architectures such as DiffServ. The focus of this paper is on transmission control protocol (TCP) traffic, and two fundamental problems related to throughput fairness are identified. First, the basic requirement of providing fair access to all users conflicts with the nature of TCP, which is fair only under certain conditions and hardly met by 802.11b WLANs. Second, short-lived TCP flows that are sensitive to losses during the early stages of TCP window growth need to be protected. To address these issues, a logical-link-control-layer algorithm that can be implemented at both access points and wireless stations is proposed. The algorithm aims at guaranteeing fair access to the medium to every user, independent of their channel conditions. At the same time, the proposed scheme protects short-lived flows, while they strive to get past the critical "small window regime." A simulation study that shows the effectiveness of the new algorithm in comparison to the standard 802.11b implementation is presented     

Recent trends in IP/NGEO satellite communication systems: transport, routing, and mobility management concerns

Non-geostationary (NGEO) satellite communication systems are seen as an attractive solution to realize the vision of anywhere, anytime pervasive access to the Internet. Their design and development have thus gained tremendous interest in the last few years. Commencing with a brief overview of general NGEO satellite configurations, this article next addresses the key technical difficulties in the development of NGEO IP-based satellite communications systems. The article discusses routing concerns, mobility management, and transport protocols with an emphasis on TCP performance in NGEO satellite networks. Some key innovations are presented. The recursive, explicit, and fair window adjustment (REFWA) scheme is presented as a solution to improve the efficiency and fairness of TCP in NGEO systems. An improvement to the REFWA scheme, REFWA plus, is also described to combat link errors in satellite environments.     

卫星TCP/ATM传输中的明确窗口指示拥塞控制策略研究

卫星ATM是近年来卫星通信领域的一个研究热点,TCP业务在卫星ATM中的传输是其中一个重要的研究课题.由于在TCP协议中主要是通过对窗口的控制来实现拥塞控制,而卫星信道传输的长延时特性大大降低了TCP层拥塞控制的效率.本文提出一种基于UBR传输的明确窗口指示拥塞控制策略,仿真结果表明在效率、公平性、VBR背景传输下的性能等各方面,取得了比目前广泛使用的Reno TCP更好的结果.并且该算法实现简单,交换机实现零信元丢失的缓冲区很小并且与TCP源连接的数目无关.     

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.     

Performance of TCP congestion control with explicit rate feedback

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.     

On the fairness characteristics of FAST TCP

Fairness of competing transmission control protocol (TCP) flows is an integral and indispensable part of transport protocol design for next-generation, high-bandwidth-delay product networks. It is not just a protocol-intrinsic property but it could also have severe impact on quality of experience (QoE). In this paper, we revisit FAST TCP fairness behavior based on a comprehensive performance evaluation study. We demonstrate that FAST TCP with proper parameter settings can always achieve fair behavior with HighSpeed TCP and Scalable TCP. We also show that this behavior is a rather robust property of the protocol concerning different traffic mix or network topology. The dynamic behavior of reaching the fair equilibrium state can be different, which is demonstrated in the paper. Our study also emphasizes the important need for finding a dynamic sensitive fairness metric for performance evaluation of transport protocols for next-generation, high-bandwidth-delay product networks.     

An early congestion feedback and rate adjustment schemes for many-to-one communication in cloud-based data center networks

Cloud data centers are playing an important role for providing many online services such as web search, cloud computing and back-end computations such as MapReduce and BigTable. In data center network, there are three basic requirements for the data center transport protocol such as high throughput, low latency and high burst tolerance. Unfortunately, conventional TCP protocols are unable to meet the requirements of data center transport protocol. One of the main practical issues of great importance is TCP Incast to occur many-to-one communication sessions in data centers, in which TCP experiences sharp degradation of throughput and higher delay. This important issue in data center networks has already attracted the researchers because of the development of cloud computing. Recently, few solutions have been proposed for improving the performance of TCP in data center networks. Among that, DCTCP is the most popular protocol in academic as well as industry areas due to its better performance in terms of throughput and latency. Although DCTCP provides significant performance improvements, there are still some defects in maintaining the queue length and throughput when the number of servers is too large. To address this problem, we propose a simple and efficient TCP protocol, namely NewDCTCP as an improvement of DCTCP in data center networks. NewDCTCP modified the congestion feedback and window adjusting schemes of DCTCP to mitigate the TCP Incast problem. Through detailed QualNet experiments, we show that NewDCTCP significantly outperforms DCTCP and TCP in terms of goodput and latency. The experimental results also demonstrate that NewDCTCP flows provide better link efficiency and fairness with respect to DCTCP.     

Fair and efficient Transmission Control Protocol access in the IEEE 802.11 infrastructure basic service set

When the stations in an IEEE 802.11 infrastructure basic service set employ Transmission Control Protocol (TCP), this exacerbates per‐flow unfair access problem. We propose a novel analytical model to approximately calculate the maximum per‐flow TCP congestion window limit that prevents packet losses at the access point buffer and therefore provides fair TCP access both in the downlink and uplink. The proposed analysis is unique in considering the effects of varying number of uplink and downlink TCP flows, differing round trip times among TCP connections and the use of delayed TCP acknowledgment (ACK) mechanism. Motivated by the findings of this theoretical analysis and simulations, we design a link layer access control block to be employed only at the access point in order to resolve the unfair access problem. The proposed link layer access control block uses congestion control and ACK filtering approach by prioritizing the access of TCP data packets of downlink flows over TCP ACK packets of uplink flows. Via simulations, we show that the proposed algorithm can provide both short‐term and long‐term fair accesses while improving channel utilization and access delay. Copyright © 2013 John Wiley & Sons, Ltd.     

W-CDMA/UMTS下一种解决TCP有效性和公平性的跨层方案

在UMTS系统中,由于MAC层的包调度引入了用户间的竞争,带来了传输层TCP有效性和公平性问题,为了解决这个问题,该文提出了一种主动ACK包转发控制策略(AAFC),它的基本思想是在基站处嵌入AAFC模块,通过MAC层和传输层之间的消息传递来控制基站处ACK包的转发,以此来保证用户间TCP流的公平性以及链路层无线资源的最充分利用。仿真结果证明了AAFC方案的有效性。     

FCA-ItswTCM:区分服务中一种公平的拥塞自适应标记算法

 本文提出了一种新的滑动窗口标记算法——公平的拥塞自适应标记算法(FCA-ItswTCM).算法近似识别TCP流和UDP流,适度区分标记,规避拥塞控制机制对公平性的影响;细粒度描述拥塞,预测拥塞,以此自适应调节各流注入黄包比例,兼顾网络拥塞状态对公平性的影响.仿真实验表明,与其他几种滑动窗口标记算法相比,FCA-ItswTCM对确保TCP流和UDP流带宽共享的公平性、提高资源利用率及系统稳定性有较好的效果.     

一种异构网络TCP拥塞控制算法

针对互联网中端对端带宽、时延和丢包率等的差异性日益加剧,导致TCP传输性能严重退化,该文提出一种链路自适应TCP拥塞控制算法(INVS)。INVS在拥塞避免阶段初期采用基于指数函数的凸窗口增长函数,以提高链路利用率;在窗口增长函数中引入了自适应增长因子实现窗口增长速率与链路状态相匹配;采用了自适应队列门限的丢包区分策略以提高无线环境下TCP的性能。性能分析和评估表明,INVS提高了TCP拥塞控制算法的吞吐量、公平性、链路利用率和RTT公平性。     

A Novel Congestion Detection Scheme in TCP Over OBS Networks

This paper introduces a novel congestion detection scheme for high-bandwidth TCP flows over optical burst switching (OBS) networks, called statistical additive increase multiplicative decrease (SAIMD). SAIMD maintains and analyzes a number of previous round-trip time (RTTs) at the TCP senders in order to identify the confidence with which a packet loss event is due to network congestion. The confidence is derived by positioning short-term RTT in the spectrum of long-term historical RTTs. The derived confidence corresponding to the packet loss is then taken in the developed policy for TCP congestion window adjustment. We will show through extensive simulation that the proposed scheme can effectively solve the false congestion detection problem and significantly outperform the conventional TCP counterparts without losing fairness. The advantages gained in our scheme are at the expense of introducing more overhead in the SAIMD TCP senders. Based on the proposed congestion control algorithm, a throughput model is formulated, and is further verified by simulation results.      

Removing redundant TCP functionalities in wired‐cum‐wireless networks with IEEE 802.11e HCCA support

The TCP was originally designed for wired networks, assuming transmission errors were negligible. Actually, any acknowledgment time‐out unconditionally triggers the congestion control mechanism, even in wireless networks in which this assumption is not valid. Consequently, in wireless networks, TCP performance significantly degrades. To avoid this degradation, this paper proposes the so‐called split TCP and UDP. In this approach, the access point splits the TCP connection and uses a customized and lighter transport protocol for the wireless segment. It takes advantage of the IEEE 802.11e Hybrid Coordination Function Controlled Channel Access (HCCA) mechanisms to remove redundant TCP functionalities. Specifically, the HCCA scheduler allows disabling of the congestion control in the wireless link. Similarly, the IEEE 802.11e error control service makes possible to eliminate TCP acknowledgments, therefore reducing the TCP protocol overhead. Finally, the usage of an HCCA scheduler permits providing fairness among the different data flows. The proposed split scheme is evaluated via extensive simulations. Results show that split TCP and User Datagram Protocol outperforms the analyzed TCP flavors—specifically designed for wireless environments—and the split TCP solution, achieving up to 95% of end‐user throughput gain. Furthermore, the proposed solution is TCP friendly because TCP flows are not degraded by the presence of flows by using this approach. Copyright © 2013 John Wiley & Sons, Ltd.     

TCP-Jersey for wireless IP communications

Improving the performance of the transmission control protocol (TCP) in wireless Internet protocol (IP) communications has been an active research area. The performance degradation of TCP in wireless and wired-wireless hybrid networks is mainly due to its lack of the ability to differentiate the packet losses caused by network congestions from the losses caused by wireless link errors. In this paper, we propose a new TCP scheme, called TCP-Jersey, which is capable of distinguishing the wireless packet losses from the congestion packet losses, and reacting accordingly. TCP-Jersey consists of two key components, the available bandwidth estimation (ABE) algorithm and the congestion warning (CW) router configuration. ABE is a TCP sender side addition that continuously estimates the bandwidth available to the connection and guides the sender to adjust its transmission rate when the network becomes congested. CW is a configuration of network routers such that routers alert end stations by marking all packets when there is a sign of an incipient congestion. The marking of packets by the CW configured routers helps the sender of the TCP connection to effectively differentiate packet losses caused by network congestion from those caused by wireless link errors. This paper describes the design of TCP-Jersey, and presents results from experiments using the NS-2 network simulator. Results from simulations show that in a congestion free network with 1% of random wireless packet loss rate, TCP-Jersey achieves 17% and 85% improvements in goodput over TCP-Westwood and TCP-Reno, respectively; in a congested network where TCP flow competes with VoIP flows, with 1% of random wireless packet loss rate, TCP-Jersey achieves 9% and 76% improvements in goodput over TCP-Westwood and TCP-Reno, respectively. Our experiments of multiple TCP flows show that TCP-Jersey maintains the fair and friendly behavior with respect to other TCP flows.     

CYRF: a theory of window-based unicast congestion control

This work presents a comprehensive theoretical framework for memoryless window-based congestion control protocols that are designed to converge to fairness and efficiency. We first derive a necessary and sufficient condition for stepwise convergence to fairness. Using this, we show how fair window increase/decrease policies can be constructed from suitable pairs of monotonically nondecreasing functions. We generalize this to smooth protocols that converge over each congestion epoch. The framework also includes a simple method for incorporating TCP-friendliness. Well-studied congestion control protocols such as TCP, GAIMD, and Binomial congestion control can be constructed using this method. Thus, we provide a common framework for the analysis of such window-based protocols. We also present two new congestion control protocols for streaming media-like applications as examples of protocol design in this framework: The first protocol, LOG, has the objective of reconciling the smoothness requirement of an application with the need for a fast dynamic response to congestion. The second protocol, SIGMOID, guarantees a minimum bandwidth for an application but behaves exactly like TCP for large windows.     

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.     

Explicit window adaptation: a method to enhance TCP performance

We study the performance of TCP in an internetwork consisting of both rate-controlled and nonrate-controlled segments. A common example of such an environment occurs when the end systems are part of IP datagram networks interconnected by a rate-controlled segment, such as an ATM network using the available bit rate (ABR) service. In the absence of congestive losses in either segment, TCP keeps increasing its window to its maximum size. Mismatch between the TCP window and the bandwidth-delay product of the network results in accumulation of large queues and possibly buffer overflows in the devices at the edges of the rate-controlled segment, causing degraded throughput and unfairness. We develop an explicit feedback scheme, called explicit window adaptation, based on modifying the receiver's advertised window in TCP acknowledgments returning to the source. The window size indicated to TCP is a function of the free buffer in the edge device. Results from simulations with a wide range of traffic scenarios show that this explicit window adaptation scheme can control the buffer occupancy efficiently at the edge device, and results in significant improvements in packet loss rate, fairness, and throughput over a packet discard policy such as random early detection (RED)     

解决FAST TCP缓存溢出相关问题的改进pacing technique算法和α参数调整算法

分析FAST TCP在缓存溢出发生时的性能,发现在缓存溢出场景中,收敛中的FAST TCP流经历严重的报文段丢失。相反,已经收敛了的FAST TCP流维持着高吞吐量和低报文段丢失概率。这种不公平是由FAST TCP缩减其窗口时的零传输率导致的。通过修改FAST TCP pacing算法,可以解决此问题。文中提出的α-adjusting算法,通过动态调整FAST TCP协议中的α参数来避免频繁的缓存溢出。通过分析ns2仿真结果,证明该算法在公平性和稳定性方面可获得令人满意的性能。     

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.