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.

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.

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

This thesis presents a model for the ACK-clock inner loop, common to virtually all Internet congestion control protocols, and analyzes the stability properties of this inner loop, as well as the stability and fairness properties of several window update mechanisms built on top of the ACK-clock. Aided by the model for the inner-loop, two new congestion control mechanisms are constructed, for wired and wireless networks. Internet traffic can be divided into two main types: TCP traffic and real-time traffic. Sending rates for TCP traffic, e.g., file-sharing, uses window-based congestion control, and adjust continuously to the network load. The sending rates for real-time traffic, e.g., voice over IP, are mostly independent of the network load. The current version of the Transmission Control Protocol (TCP) results in large queueing delays at bottlenecks, and poor quality for real-time applications that share a bottleneck link with TCP. The first contribution is a new model for the dynamic relationship between window sizes, sending rates, and queue sizes. This system, with window sizes as inputs, and queue sizes as outputs, is the inner loop at the core of window-based congestion control. The new model unifies two models that have been widely used in the literature. The dynamics of this system, including the static gain and the time constant, depend on the amount of cross traffic which is not subject to congestion control. The model is validated using ns-2 simulations, and it is shown that the system is stable. For moderate cross traffic, the system convergence time is a couple of roundtrip times. When introducing a new congestion control protocol, one important question is how flows using different protocols share resources. The second contribution is an analysis of the fairness when a flow using TCP Westwood+ is introduced in a network that is also used by a TCP New Reno flow. It is shown that the sharing of capacity depends on the buffer size at the bottleneck link. With a buffer size matching the bandwidth-delay product, both flows get equal shares. If the buffer size is smaller, Westwood+ gets a larger share. In the limit of zero buffering, it gets all the capacity. If the buffer size is larger, New Reno gets a larger share. In the limit of very large buffers, it gets 3/4 of the capacity. The third contribution is a new congestion control mechanism, maintaining small queues. The overall control structure is similar to the combination of TCP with Active Queue Management (AQM) and explicit congestion notification, where routers mark some packets according to a probability which depends on the queue size. The key ideas are to take advantage of the stability of the inner loop, and to use control laws for setting and reacting to packet marks that result in more frequent feedback than with AQM. Stability analysis for the single flow, single bottleneck topology gives a simple stability condition, which can be used to guide tuning. Simulations, both of the fluid-flow differential equations, and in the ns-2 packet simulator, show that the protocol maintains small queues. The simulations also indicate that tuning, using a single control parameter per link, is fairly easy. The final contribution is a split-connection scheme for downloads to a mobile terminal. A wireless mobile terminal requests a file from a web server, via a proxy. During the file transfer, the Radio Network Controller (RNC) informs the proxy about bandwidth changes over the radio channel, and the current RNC queue length. A novel control mechanism in the proxy uses this information to adjust the window size. In simulation studies, including one based on detailed radio-layer simulations, both the user response time and the link utilization are improved, compared TCP New Reno, Eifel and Snoop, both for a dedicated channel, and for the shared channel in High-Speed Downlink Packet Access. / QC 20100830

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.

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.