In wireless sensor networks (WSN), nodes have very limited power due to hardware constraints. Packet losses and retransmissions resulting from congestion cost precious energy and shorten the lifetime of sensor nodes. This problem motivates the need for congestion control mechanisms in WSN. In this thesis, an observation of multiple non-empty queues in sensor networks is first reported. Other aspects affected by congestion like queue length, delay and packet loss are also studied. The simulation results show that the number of occupied queues along a path can be used to detect congestion. Based on the above result, a congestion control scheme for the transport layer is proposed in this thesis. It is composed of three parts: (i) congestion detection by tracking the number of non-empty queues; (ii) On-demand midway non-binary explicit congestion notification (CN) feedback; and (iii) Adaptive rate control based on additive increase and multiplicative decrease (AIMD). This scheme has been implemented in ns2. Extensive simulations have been conducted to evaluate it. Results show that it works well in mitigating and avoiding congestion and achieves good performance in terms of energy dissipation, latency and transmission effciency.

No description available.

Congestion control in wireline networks has been studied extensively since the seminal work by Mazumdar et al in 1998. It is well known that this global optimization problem can be implemented in a distributed manner. Stability and fairness are two main design objectives of congestion control mechanisms. Most literatures make the assumption that the number of flows is fixed in the network and each flow has infinite backlog for transfer in developing congestion control schemes. However, this assumption may not hold in reality. Thus, there is a need to study congestion control algorithm in the presence of dynamic flows. It is only until recently that short-lived flows have been taken into account. In this thesis, we study utility maximization problems for networks with dynamic flows. In particular, we consider the case where each class of flows arrives according to a Poisson process and has a length given by a certain distribution. The goal is to maximize the long-term expected system utility, which is a function of the number of flows and the rate (identical within a given class) allocated to each flow. Our investigation shows that, as long as the average work brought by the arrival processes is strictly within the network stability region, the fairness and stability issues are independent. While stability can be guaranteed by, for example, a FIFO policy, utility maximization becomes an unconstrained optimization. We also provide a queueing interpretation of this seemingly surprising result and show that not all utility functions make sense under dynamic flows. Finally, we use simulation results to show that our algorithm indeed maximizes the expected system utility.

A cluster-based router is a new router architecture that is composed of a cluster of commodity processing nodes interconnected by a high-speed and low-latency network. It inherits packet processing extensibility from the software router, and forwarding performance scalability from clustering. In this thesis, we describe a prototype cluster-based router, including the design of the cluster-based router architecture and the addressing of critical issues such as the design of a highly efficient communication layer, reduction of operating system overheads, buffer recycling and packet packing. By experimental evaluation, we expose its forwarding capacity scalability and latency variance. We also evaluate and analyze the potential hardware bottlenecks of its commodity processing nodes, and present the correlation between the reception and transmission capabilities of an individual port as well as ports on the same bus. We propose an adaptive scheduling mechanism based on system state information to manage the adverse effect of this correlation on the router performance. We also investigate internal congestion in the cluster-based router. To manage the internal congestion, we propose two backward explicit congestion notification schemes: a novel queue scheduling method and an optimal utility-based scheme. We show the effectiveness of these schemes either by ns-3 simulation, experimental evaluation, or both. We also analyze the stability of the optimal utility-based BECN internal congestion control scheme through theoretical proof, simulation and experimental evaluation.

Video conferencing applications have significantly changed the way in which people communicate over the Internet. Web real-time communication (WebRTC), drafted by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF), has added new functionality to web browsers, allowing audio/video calls between browsers without the need to install any video telephony applications. The Google Congestion Control (GCC) algorithm has been proposed as WebRTC’s receiver congestion control mechanism, but its performance is limited due to using a fixed incoming rate decrease factor, known as an alpha (α). In this thesis, we have proposed a dynamic alpha model to reduce the receiving bandwidth estimate during overuse, as indicated by the overuse detector. Experiments using our specific testbed show that our proposed model achieves a higher incoming rate and a lower Round-Trip Time while slightly increasing the packet loss rate in some cases compared to fixed alpha model. Our mathematical model proves that it is necessary to use an adaptive alpha α as the receiver side controller. The experimental results show improvement in the term of incoming rate, Round-Trip Time, and packet fraction loss rate in some cases. Our model increases the amount of incoming rate and decreases Round-Trip Time and fraction loss.

In this work, we focus on congestion control mechanisms in Transmission Control Protocol (TCP) for emerging very-high bandwidth-delay product networks and suggest several congestion control schemes for parallel and single-flow TCP. Recently, several high-speed TCP proposals have been suggested to overcome the limited throughput achievable by single-flow TCP by modifying its congestion control mechanisms. In the meantime, users overcome the throughput limitations in high bandwidth-delay product networks by using multiple parallel TCP flows, without modifying TCP itself. However, the evident lack of fairness between the high-speed TCP proposals (or parallel TCP) and existing standard TCP has increasingly become an issue. In many scenarios where flows require high throughput, such as grid computing or content distribution networks, often multiple connections go to the same or nearby destinations and tend to share long portions of paths (and bottlenecks). In such cases benefits can be gained by sharing congestion information. To take advantage of this additional information, we first propose a collaborative congestion control scheme for parallel TCP flows. Although the use of parallel TCP flows is an easy and effective way for reliable high-speed data transfer, parallel TCP flows are inherently unfair with respect to single TCP flows. In this thesis we propose, implement, and evaluate a natural extension for aggregated aggressiveness control in parallel TCP flows. To improve the effectiveness of single TCP flows over high bandwidth-delay product networks without causing fairness problems, we suggest a new TCP congestion control scheme that effectively and fairly utilizes high bandwidth-delay product networks by adaptively controlling the flow’s aggressiveness according to network situations using a competition detection mechanism. We argue that competition detection is more appropriate than congestion detection or bandwidth estimation. We further extend the adaptive aggressiveness control mechanism and the competition detection mechanism from single flows to parallel flows. In this way we achieve adaptive aggregated aggressiveness control. Our evaluations show that the resulting implementation is effective and fair. As a result, we show that single or parallel TCP flows in end-hosts can achieve high performance over emerging high bandwidth-delay product networks without requiring special support from networks or modifications to receivers.

Despite the tremendous bandwidth increase in 3rd generation (3G) Broadband Wireless Networks (BWNs) such as Universal Mobile Telecommunication System (UMTS), maintaining the mobile users’ Quality of Service (QoS) requirements while maximizing the network operators’ revenues is still a challenging issue. Moreover, spatial distribution of network traffic has a negative impact on the overall network performance where network resources are overutilized in parts of the network coverage area while such resources are underutilized in other network coverage areas. Therefore, network congestion and traffic imbalance become inevitable. Hence, efficient Radio Resource Management (RRM) techniques which release congestion and balance network traffic are of utmost need for the success of such wireless cellular systems. Congestion control and load balancing in BWNs are, however, challenging tasks due to the complexity of these systems and the multiple dimensions that need to be taken into consideration. Examples of such issues include the diverse QoS requirements of the supported multimedia services, the interference level in the system, which vary the mobile users and base stations allocated transmission powers and transmission rates to guarantee certain QoS levels during the lifetime of mobile users connections. In this thesis, we address the problem of congestion control and load balancing in BWNs and propose efficient network coverage adaptation solution in order to deal with these issues, and hence enhance the QoS support in these systems. Specifically, we propose a directional coverage adaptation framework for BWNs. The framework is designed to dynamically vary the coverage level of network cells to release system congestion and balance traffic load by forcing mobile users handoff from a loaded cell to its nearby lightly loaded cell. The framework consists of three related components, namely directional coverage adaptation module, congestion control and load balancing protocol, and QoS provisioning module. These components interact with each other to release system congestion, balance network load, maximize network resource utilization, while maintaining the required QoS parameters for individual mobile users. / Thesis (Ph.D, Electrical & Computer Engineering) -- Queen's University, 2009-04-24 12:15:54.582

For any congestion control mechanisms, the most fundamental design objectives are stability and scalability. However, achieving both properties are very challenging in such a heterogeneous environment as the Internet. From the end-users' perspective, heterogeneity is due to the fact that different flows have different routing paths and therefore different communication delays, which can significantly affect stability of the entire system. In this work, we successfully address this problem by first proving a sufficient and necessary condition for a system to be stable under arbitrary delay. Utilizing this result, we design a series of practical congestion control protocols (MKC and JetMax) that achieve stability regardless of delay as well as many additional appealing properties. From the routers' perspective, the system is heterogeneous because the incoming traffic is a mixture of short- and long-lived, TCP and non-TCP flows. This imposes a severe challenge on traditional buffer sizing mechanisms, which are derived using the simplistic model of a single or multiple synchronized long-lived TCP flows. To overcome this problem, we take a control-theoretic approach and design a new intelligent buffer sizing scheme called Adaptive Buffer Sizing (ABS), which based on the current incoming traffic, dynamically sets the optimal buffer size under the target performance constraints. Our extensive simulation results demonstrate that ABS exhibits quick responses to changes of traffic load, scalability to a large number of incoming flows, and robustness to generic Internet traffic.

We investigate the issues in making a delay-based protocol adaptive to heterogeneous environments. We assess and address the problems a delay-based protocol faces when competing with a loss-based protocol such as TCP. We investigate if noise and variability in delay measurements in environments such as cable and ADSL access networks impact the delay-based protocol behavior significantly. We investigate these issues in the context of incremental deployment of a new delay-based protocol, PERT. We propose design modifications to PERT to compete with the TCP flavor SACK. We show through simulations and real network experiments that, with the proposed changes, PERT experiences lower drop rates than SACK and leads to lower overall drop rates with different mixes of PERT and SACK protocols. Delay-based protocols, being less aggressive, have problems in fully utilizing a highspeed link while operating alone. We show that a single PERT flow can fully utilize a high-speed, high-delay link. We performed several experiments with diverse parameters and simulated numerous scenarios using ns-2. The results from simulations indicate that PERT can adapt to heterogeneous networks and can operate well in an environment of heterogeneous protocols and other miscellaneous scenarios like wireless networks (in the presence of channel errors). We also show that proposed changes retain the desirable properties of PERT such as low loss rates and fairness when operating alone. To see how the protocol performs with the real-world traffic, the protocol has also been implemented in the Linux kernel and tested through experiments on live networks, by measuring the throughput and losses between nodes in our lab at TAMU and different machines at diverse location across the globe on the planet-lab. The results from simulations indicate that PERT can compete with TCP in diverse environments and provides benefits as it is incrementally deployed. Results from real-network experiments strengthen this claim as PERT shows similar behavior with the real-world traffic.

The proliferation and universal adoption of the Internet has made it become the key information transport platform of our time. Congestion occurs when resource demands exceed the capacity, which results in poor performance in the form of low network utilization and high packet loss rate. The goal of congestion control mechanisms is to use the network resources as efficiently as possible. The research work in this thesis is centered on finding ways to address these types of problems and provide guidelines for predicting and controlling network performance, through the use of suitable mathematical tools and control analysis. The first congestion collapse in the Internet was observed in 1980's. To solve the problem, Van Jacobson proposed the Transmission Control Protocol (TCP) congestion control algorithm based on the Additive Increase and Multiplicative Decrease (AIMD) mechanism in 1988. To be effective, a congestion control mechanism must be paired with a congestion detection scheme. To detect and distribute network congestion indicators fairly to all on-going flows, Active Queue Management (AQM), e.g., the Random Early Detection (RED) queue management scheme has been developed to be deployed in the intermediate nodes. The currently dominant AIMD congestion control, coupled with the RED queue in the core network, has been acknowledged as one of the key factors to the overwhelming success of the Internet. In this thesis, the AIMD/RED system, based on the fluid-flow model, is systematically studied. In particular, we concentrate on the system modeling, stability analysis and bounds estimates. We first focus on the stability and fairness analysis of the AIMD/RED system with a single bottleneck. Then, we derive the theoretical estimates for the upper and lower bounds of homogeneous and heterogeneous AIMD/RED systems with feedback delays and further discuss the system performance when it is not asymptotically stable. Last, we develop a general model for a class of multiple-bottleneck networks and discuss the stability properties of such a system. Theoretical and simulation results presented in this thesis provide insights for in-depth understanding of AIME/RED system and help predict and control the system performance for the Internet with higher data rate links multiplexed with heterogeneous flows.