Active Queue Management for Fair Resource Allocation in Wireless Networks

This paper investigates the interaction between end-to-end flow control and MAC-layer scheduling on wireless links. We consider a wireless network with multiple users receiving information from a common access point; each user suffers fading, and a scheduler allocates the channel based on channel quality,but subject to fairness and latency considerations. We show that the fairness property of the scheduler is compromised by the transport layer flow control of TCP New Reno. We provide a receiver-side control algorithm, CLAMP, that remedies this situation. CLAMP works at a receiver to control a TCP sender by setting the TCP receiver's advertised window limit, and this allows the scheduler to allocate bandwidth fairly between the users

BaseStation Assisted TCP: A Simple Way to Improve Wireless TCP

Abstract. In recent years, extensive research effort has been devoted to TCP congestion control in hybrid wired-wireless networks. A general agreement is that the TCP sender should respond differently to wireless losses and disconnection, i.e., not slow down as drastically as for congestion losses. Thus, research focus for wireless TCP congestion control is the discrimination between the wireless inherent packet losses and the network congestion packet losses in wired network. In addition, researchers attempt to detect temporary or lengthy wireless disconnection. This paper proposes a simple but novel strategy, dubbed BSA-TCP (Base Station Assisted TCP), (1) to accurately discriminate wireless losses from wired network congestion losses and (2) to detect and notify a TCP sender about wireless disconnections. The key distinctive feature of the proposed scheme is its general use for most issues at stake for TCP over wireless: loss discrimination, wireless disconnection and handoffs. It also circumvents the asymmetric problem that acknowledgements might follow different paths from those of data packets. Such asymmetry problem is common to mechanisms that buffer and retransmit wireless lost data packets locally at the base station. The proposed method also addresses energy efficiency

TCP performance enhancement in wireless networks via adaptive congestion control and active queue management

The transmission control protocol (TCP) exhibits poor performance when used in error-prone wireless networks. Remedy to this problem has been an active research area. However, a widely accepted and adopted solution is yet to emerge. Difficulties of an acceptable solution lie in the areas of compatibility, scalability, computational complexity and the involvement of intermediate routers and switches. This dissertation rexriews the current start-of-the-art solutions to TCP performance enhancement, and pursues an end-to-end solution framework to the problem. The most noticeable cause of the performance degradation of TCP in wireless networks is the higher packet loss rate as compared to that in traditional wired networks. Packet loss type differentiation has been the focus of many proposed TCP performance enhancement schemes. Studies conduced by this dissertation research suggest that besides the standard TCP\u27s inability of discriminating congestion packet losses from losses related to wireless link errors, the standard TCP\u27s additive increase and multiplicative decrease (AIMD) congestion control algorithm itself needs to be redesigned to achieve better performance in wireless, and particularly, high-speed wireless networks. This dissertation proposes a simple, efficient, and effective end-to-end solution framework that enhances TCP\u27s performance through techniques of adaptive congestion control and active queue management. By end-to-end, it means a solution with no requirement of routers being wireless-aware or wireless-specific . TCP-Jersey has been introduced as an implementation of the proposed solution framework, and its performance metrics have been evaluated through extensive simulations. TCP-Jersey consists of an adaptive congestion control algorithm at the source by means of the source\u27s achievable rate estimation (ARE) —an adaptive filter of packet inter-arrival times, a congestion indication algorithm at the links (i.e., AQM) by means of packet marking, and a effective loss differentiation algorithm at the source by careful examination of the congestion marks carried by the duplicate acknowledgment packets (DUPACK). Several improvements to the proposed TCP-Jersey have been investigated, including a more robust ARE algorithm, a less computationally intensive threshold marking algorithm as the AQM link algorithm, a more stable congestion indication function based on virtual capacity at the link, and performance results have been presented and analyzed via extensive simulations of various network configurations. Stability analysis of the proposed ARE-based additive increase and adaptive decrease (AJAD) congestion control algorithm has been conducted and the analytical results have been verified by simulations. Performance of TCP-Jersey has been compared to that of a perfect , but not practical, TCP scheme, and encouraging results have been observed. Finally the framework of the TCP-Jersey\u27s source algorithm has been extended and generalized for rate-based congestion control, as opposed to TCP\u27s window-based congestion control, to provide a design platform for applications, such as real-time multimedia, that do not use TCP as transport protocol yet do need to control network congestion as well as combat packet losses in wireless networks. In conclusion, the framework architecture presented in this dissertation that combines the adaptive congestion control and active queue management in solving the TCP performance degradation problem in wireless networks has been shown as a promising answer to the problem due to its simplistic design philosophy complete compatibility with the current TCP/IP and AQM practice, end-to-end architecture for scalability, and the high effectiveness and low computational overhead. The proposed implementation of the solution framework, namely TCP-Jersey is a modification of the standard TCP protocol rather than a completely new design of the transport protocol. It is an end-to-end approach to address the performance degradation problem since it does not require split mode connection establishment and maintenance using special wireless-aware software agents at the routers. The proposed solution also differs from other solutions that rely on the link layer error notifications for packet loss differentiation. The proposed solution is also unique among other proposed end-to-end solutions in that it differentiates packet losses attributed to wireless link errors from congestion induced packet losses directly from the explicit congestion indication marks in the DUPACK packets, rather than inferring the loss type based on packet delay or delay jitter as in many other proposed solutions; nor by undergoing a computationally expensive off-line training of a classification model (e.g., HMM), or a Bayesian estimation/detection process that requires estimations of a priori loss probability distributions of different loss types. The proposed solution is also scalable and fully compatible to the current practice in Internet congestion control and queue management, but with an additional function of loss type differentiation that effectively enhances TCP\u27s performance over error-prone wireless networks. Limitations of the proposed solution architecture and areas for future researches are also addressed

Enhancing TCP Performance in Mobile Ad Hoc Network Using Explicit Link Failure Notification (ELFN)

The dynamics and the unpredictable behaviour of a wireless mobile ad hoc network results in the hindrance of providing adequate reliability to network connections. Frequent route changes in the network relatively introduce incessant link failures which eventually degrade TCP performance considerably. In this research, we are going to study the potential improvement of TCP performance when Explicit Link Failure Notification is implemented as opposed to the standard TCP mechanism. ELFN modifies the ‘slow start’ mechanism that is used in standard TCP so that the throughput achieved from the network can be maximized

Cooperative End-to-end Congestion Control in Heterogeneous Wireless Networks

Sharing the resources of multiple wireless networks with overlapped coverage areas has a potential of improving the transmission throughput. However, in the existing frameworks, the improvement cannot be achieved in congestion scenarios because of independent congestion control procedures among the end-to-end paths. Although various network characteristics make the congestion control complex, this variety can be useful in congestion avoidance if the networks cooperate with each other. When congestion happens in an end-to-end path, it is inevitable to have a packet transmission rate less than the minimum requested rate due to congestion window size adjustments. Cooperation among networks can help to avoid this problem for better service quality. When congestion is predicted for one path, some of the on-going packets can be sent over other paths instead of the congested path. In this way, the traffic can be shifted from a congested network to others, and the overall transmission throughput does not degrade in a congestion scenario. However, cooperation is not always advantageous since the throughput of cooperative transmission in an uncongested scenario can be less than that of non-cooperative transmission due to cooperation costs such as cooperation setup time, additional signalling for cooperation, and out-of-order packet reception. In other words, a trade-off exists between congestion avoidance and cooperation cost. Thus, cooperation should be triggered only when it is beneficial according to congestion level measurements. In this research, our aim is to develop an efficient cooperative congestion control scheme for a heterogeneous wireless environment. To this end, a cooperative congestion control algorithm is proposed, in which the state of an end-to-end path is provided at the destination terminal by measuring the queuing delay and estimating the congestion level. The decision on when to start/stop cooperation is made based on the network characteristics, instantaneous traffic condition, and the requested quality of service (QoS). Simulation results demonstrate the throughput improvement of the proposed scheme over non-cooperative congestion control.1 yea

Improved algorithms for TCP congestion control

Reliable and efficient data transfer on the Internet is an important issue. Since late 70’s the protocol responsible for that has been the de facto standard TCP, which has proven to be successful through out the years, its self-managed congestion control algorithms have retained the stability of the Internet for decades. However, the variety of existing new technologies such as high-speed networks (e.g. fibre optics) with high-speed long-delay set-up (e.g. cross-Atlantic links) and wireless technologies have posed lots of challenges to TCP congestion control algorithms. The congestion control research community proposed solutions to most of these challenges. This dissertation adds to the existing work by: firstly tackling the highspeed long-delay problem of TCP, we propose enhancements to one of the existing TCP variants (part of Linux kernel stack). We then propose our own variant: TCP-Gentle. Secondly, tackling the challenge of differentiating the wireless loss from congestive loss in a passive way and we propose a novel loss differentiation algorithm which quantifies the noise in packet inter arrival times and use this information together with the span (ratio of maximum to minimum packet inter arrival times) to adapt the multiplicative decrease factor according to a predefined logical formula. Finally, extending the well-known drift model of TCP to account for wireless loss and some hypothetical cases (e.g. variable multiplicative decrease), we have undertaken stability analysis for the new version of the model

Anakyzing the performance of Active Queue Management Algorithms

Congestion is an important issue which researchers focus on in the Transmission Control Protocol (TCP) network environment. To keep the stability of the whole network, congestion control algorithms have been extensively studied. Queue management method employed by the routers is one of the important issues in the congestion control study. Active queue management (AQM) has been proposed as a router-based mechanism for early detection of congestion inside the network. In this paper we analyzed several active queue management algorithms with respect to their abilities of maintaining high resource utilization, identifying and restricting disproportionate bandwidth usage, and their deployment complexity. We compare the performance of FRED, BLUE, SFB, and CHOKe based on simulation results, using RED and Drop Tail as the evaluation baseline. The characteristics of different algorithms are also discussed and compared. Simulation is done by using Network Simulator(NS2) and the graphs are drawn using X- graph.Comment: 19 Pages, IJCNC Journal 201

End-to-End Resilience Mechanisms for Network Transport Protocols

The universal reliance on and hence the need for resilience in network communications has been well established. Current transport protocols are designed to provide fixed mechanisms for error remediation (if any), using techniques such as ARQ, and offer little or no adaptability to underlying network conditions, or to different sets of application requirements. The ubiquitous TCP transport protocol makes too many assumptions about underlying layers to provide resilient end-to-end service in all network scenarios, especially those which include significant heterogeneity. Additionally the properties of reliability, performability, availability, dependability, and survivability are not explicitly addressed in the design, so there is no support for resilience. This dissertation presents considerations which must be taken in designing new resilience mechanisms for future transport protocols to meet service requirements in the face of various attacks and challenges. The primary mechanisms addressed include diverse end-to-end paths, and multi-mode operation for changing network conditions

Improving TCP performance over heterogeneous networks : The investigation and design of End to End techniques for improving TCP performance for transmission errors over heterogeneous data networks.

Transmission Control Protocol (TCP) is considered one of the most important protocols in the Internet. An important mechanism in TCP is the congestion control mechanism which controls TCP sending rate and makes TCP react to congestion signals. Nowadays in heterogeneous networks, TCP may work in networks with some links that have lossy nature (wireless networks for example). TCP treats all packet loss as if they were due to congestion. Consequently, when used in networks that have lossy links, TCP reduces sending rate aggressively when there are transmission (non-congestion) errors in an uncongested network. One solution to the problem is to discriminate between errors; to deal with congestion errors by reducing TCP sending rate and use other actions for transmission errors. In this work we investigate the problem and propose a solution using an end-to-end error discriminator. The error discriminator will improve the current congestion window mechanism in TCP and decide when to cut and how much to cut the congestion window. We have identified three areas where TCP interacts with drops: congestion window update mechanism, retransmission mechanism and timeout mechanism. All of these mechanisms are part of the TCP congestion control mechanism. We propose changes to each of these mechanisms in order to allow TCP to cope with transmission errors. We propose a new TCP congestion window action (CWA) for transmission errors by delaying the window cut decision until TCP receives all duplicate acknowledgments for a given window of data (packets in flight). This will give TCP a clear image about the number of drops from this window. The congestion window size is then reduced only by number of dropped packets. Also, we propose a safety mechanism to prevent this algorithm from causing congestion to the network by using an extra congestion window threshold (tthresh) in order to save the safe area where there are no drops of any kind. The second algorithm is a new retransmission action to deal with multiple drops from the same window. This multiple drops action (MDA) will prevent TCP from falling into consecutive timeout events by resending all dropped packets from the same window. A third algorithm is used to calculate a new back-off policy for TCP retransmission timeout based on the network¿s available bandwidth. This new retransmission timeout action (RTA) helps relating the length of the timeout event with current network conditions, especially with heavy transmission error rates. The three algorithms have been combined and incorporated into a delay based error discriminator. The improvement of the new algorithm is measured along with the impact on the network in terms of congestion drop rate, end-to-end delay, average queue size and fairness of sharing the bottleneck bandwidth. The results show that the proposed error discriminator along with the new actions toward transmission errors has increased the performance of TCP. At the same time it has reduced the load on the network compared to existing error discriminators. Also, the proposed error discriminator has managed to deliver excellent fairness values for sharing the bottleneck bandwidth. Finally improvements to the basic error discriminator have been proposed by using the multiple drops action (MDA) for both transmission and congestion errors. The results showed improvements in the performance as well as decreases in the congestion loss rates when compared to a similar error discriminator.Ministry of Higher Education and King Saud University in Saudi Arabia

Transport Control Protocol (TCP) over Optical Burst Switched Networks

Transport Control Protocol (TCP) is the dominant protocol in modern communication networks, in which the issues of reliability, flow, and congestion control must be handled efficiently. This thesis studies the impact of the next-generation bufferless optical burst-switched (OBS) networks on the performance of TCP congestion-control implementations (i.e., dropping-based, explicit-notification-based, and delay-based). The burst contention phenomenon caused by the buffer-less nature of OBS occurs randomly and has a negative impact on dropping-based TCP since it causes a false indication of network congestion that leads to improper reaction on a burst drop event. In this thesis we study the impact of these random burst losses on dropping-based TCP throughput. We introduce a novel congestion control scheme for TCP over OBS networks, called Statistical Additive Increase Multiplicative Decrease (SAIMD). SAIMD maintains and analyzes a number of previous round trip times (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 to account by the policy developed for TCP congestion-window adjustment. For explicit-notification TCP, we propose a new TCP implementation over OBS networks, called TCP with Explicit Burst Loss Contention Notification (TCP-BCL). We examine the throughput performance of a number of representative TCP implementations over OBS networks, and analyze the TCP performance degradation due to the misinterpretation of timeout and packet-loss events. We also demonstrate that the proposed TCP-BCL scheme can counter the negative effect of OBS burst losses and is superior to conventional TCP architectures in OBS networks. For delay-based TCP, we observe that this type of TCP implementation cannot detect network congestion when deployed over typical OBS networks since RTT fluctuations are minor. Also, delay-based TCP can suffer from falsely detecting network congestion when the underlying OBS network provides burst retransmission and/or deflection. Due to the fact that burst retransmission and deflection schemes introduce additional delays for bursts that are retransmitted or deflected, TCP cannot determine whether this sudden delay is due to network congestion or simply to burst recovery at the OBS layer. In this thesis we study the behaviour of delay-based TCP Vegas over OBS networks, and propose a version of threshold-based TCP Vegas that is suitable for the characteristics of OBS networks. The threshold-based TCP Vegas is able to distinguish increases in packet delay due to network congestion from burst contention at low traffic loads. The evolution of OBS technology is highly coupled with its ability to support upper-layer applications. Without fully understanding the burst transmission behaviour and the associated impact on the TCP congestion-control mechanism, it will be difficult to exploit the advantages of OBS networks fully