Enhancing QoS provisioning and granularity in next generation internet

Next Generation IP technology has the potential to prevail, both in the access and in the core networks, as we are moving towards a multi-service, multimedia and high-speed networking environment. Many new applications, including the multimedia applications, have been developed and deployed, and demand Quality of Service (QoS) support from the Internet, in addition to the current best effort service. Therefore, QoS provisioning techniques in the Internet to guarantee some specific QoS parameters are more a requirement than a desire. Due to the large amount of data flows and bandwidth demand, as well as the various QoS requirements, scalability and fine granularity in QoS provisioning are required. In this dissertation, the end-to-end QoS provisioning mechanisms are mainly studied, in order to provide scalable services with fine granularity to the users, so that both users and network service providers can achieve more benefits from the QoS provisioned in the network. To provide the end-to-end QoS guarantee, single-node QoS provisioning schemes have to be deployed at each router, and therefore, in this dissertation, such schemes are studied prior to the study of the end-to-end QoS provisioning mechanisms. Specifically, the effective sharing of the output bandwidth among the large amount of data flows is studied, so that fairness in the bandwidth allocation among the flows can be achieved in a scalable fashion. A dual-rate grouping architecture is proposed in this dissertation, in which the granularity in rate allocation can be enhanced, while the scalability of the one-rate grouping architecture is still maintained. It is demonstrated that the dual-rate grouping architecture approximates the ideal per-flow based PFQ architecture better than the one-rate grouping architecture, and provides better immunity capability. On the end-to-end QoS provisioning, a new Endpoint Admission Control scheme for Diffserv networks, referred to as Explicit Endpoint Admission Control (EEAC), is proposed, in which the admission control decision is made by the end hosts based on the end-to-end performance of the network. A novel concept, namely the service vector, is introduced, by which an end host can choose different services at different routers along its data path. Thus, the proposed service provisioning paradigm decouples the end-to-end QoS provisioning from the service provisioning at each router, and the end-to-end QoS granularity in the Diffserv networks can be enhanced, while the implementation complexity of the Diffserv model is maintained. Furthermore, several aspects of the implementation of the EEAC and service vector paradigm, referred to as EEAC-SV, in the Diffserv architecture are also investigated. The performance analysis and simulation results demonstrate that the proposed EEAC-SV scheme, not only increases the benefit to the service users, but also enhances the benefit to the network service provider in terms of network resource utilization. The study also indicates that the proposed EEAC-SV scheme can provide a compatible and friendly networking environment to the conventional TCP flows, and the scheme can be deployed in the current Internet in an incremental and gradual fashion

Application-Oriented Flow Control: Fundamentals, Algorithms and Fairness

This paper is concerned with flow control and resource allocation problems in computer networks in which real-time applications may have hard quality of service (QoS) requirements. Recent optimal flow control approaches are unable to deal with these problems since QoS utility functions generally do not satisfy the strict concavity condition in real-time applications. For elastic traffic, we show that bandwidth allocations using the existing optimal flow control strategy can be quite unfair. If we consider different QoS requirements among network users, it may be undesirable to allocate bandwidth simply according to the traditional max-min fairness or proportional fairness. Instead, a network should have the ability to allocate bandwidth resources to various users, addressing their real utility requirements. For these reasons, this paper proposes a new distributed flow control algorithm for multiservice networks, where the application's utility is only assumed to be continuously increasing over the available bandwidth. In this, we show that the algorithm converges, and that at convergence, the utility achieved by each application is well balanced in a proportionally (or max-min) fair manner

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

Explicit congestion control algorithms for available bit rate services in asynchronous transfer mode networks

Congestion control of available bit rate (ABR) services in asynchronous transfer mode (ATM) networks has been the recent focus of the ATM Forum. The focus of this dissertation is to study the impact of queueing disciplines on ABR service congestion control, and to develop an explicit rate control algorithm. Two queueing disciplines, namely, First-In-First-Out (FIFO) and per-VC (virtual connection) queueing, are examined. Performance in terms of fairness, throughput, cell loss rate, buffer size and network utilization are benchmarked via extensive simulations. Implementation complexity analysis and trade-offs associated with each queueing implementation are addressed. Contrary to the common belief, our investigation demonstrates that per-VC queueing, which is costlier and more complex, does not necessarily provide any significant improvement over simple FIFO queueing. A new ATM switch algorithm is proposed to complement the ABR congestion control standard. The algorithm is designed to work with the rate-based congestion control framework recently recommended by the ATM Forum for ABR services. The algorithm\u27s primary merits are fast convergence, high throughput, high link utilization, and small buffer requirements. Mathematical analysis is done to show that the algorithm converges to the max-min fair allocation rates in finite time, and the convergence time is proportional to the distinct number of fair allocations and the round-trip delays in the network. At the steady state, the algorithm operates without causing any oscillations in rates. The algorithm does not require any parameter tuning, and proves to be very robust in a large ATM network. The impact of ATM switching and ATM layer congestion control on the performance of TCP/IP traffic is studied and the results are presented. The study shows that ATM layer congestion control improves the performance of TCP/IP traffic over ATM, and implementing the proposed switch algorithm drastically reduces the required switch buffer requirements. In order to validate claims, many benchmark ATM networks are simulated, and the performance of the switch is evaluated in terms of fairness, link utilization, response time, and buffer size requirements. In terms of performance and complexity, the algorithm proposed here offers many advantages over other proposed algorithms in the literature

Datacenter Traffic Control: Understanding Techniques and Trade-offs

Datacenters provide cost-effective and flexible access to scalable compute and storage resources necessary for today's cloud computing needs. A typical datacenter is made up of thousands of servers connected with a large network and usually managed by one operator. To provide quality access to the variety of applications and services hosted on datacenters and maximize performance, it deems necessary to use datacenter networks effectively and efficiently. Datacenter traffic is often a mix of several classes with different priorities and requirements. This includes user-generated interactive traffic, traffic with deadlines, and long-running traffic. To this end, custom transport protocols and traffic management techniques have been developed to improve datacenter network performance. In this tutorial paper, we review the general architecture of datacenter networks, various topologies proposed for them, their traffic properties, general traffic control challenges in datacenters and general traffic control objectives. The purpose of this paper is to bring out the important characteristics of traffic control in datacenters and not to survey all existing solutions (as it is virtually impossible due to massive body of existing research). We hope to provide readers with a wide range of options and factors while considering a variety of traffic control mechanisms. We discuss various characteristics of datacenter traffic control including management schemes, transmission control, traffic shaping, prioritization, load balancing, multipathing, and traffic scheduling. Next, we point to several open challenges as well as new and interesting networking paradigms. At the end of this paper, we briefly review inter-datacenter networks that connect geographically dispersed datacenters which have been receiving increasing attention recently and pose interesting and novel research problems.Comment: Accepted for Publication in IEEE Communications Surveys and Tutorial

User-Centric Quality of Service Provisioning in IP Networks

The Internet has become the preferred transport medium for almost every type of communication, continuing to grow, both in terms of the number of users and delivered services. Efforts have been made to ensure that time sensitive applications receive sufficient resources and subsequently receive an acceptable Quality of Service (QoS). However, typical Internet users no longer use a single service at a given point in time, as they are instead engaged in a multimedia-rich experience, comprising of many different concurrent services. Given the scalability problems raised by the diversity of the users and traffic, in conjunction with their increasing expectations, the task of QoS provisioning can no longer be approached from the perspective of providing priority to specific traffic types over coexisting services; either through explicit resource reservation, or traffic classification using static policies, as is the case with the current approach to QoS provisioning, Differentiated Services (Diffserv). This current use of static resource allocation and traffic shaping methods reveals a distinct lack of synergy between current QoS practices and user activities, thus highlighting a need for a QoS solution reflecting the user services. The aim of this thesis is to investigate and propose a novel QoS architecture, which considers the activities of the user and manages resources from a user-centric perspective. The research begins with a comprehensive examination of existing QoS technologies and mechanisms, arguing that current QoS practises are too static in their configuration and typically give priority to specific individual services rather than considering the user experience. The analysis also reveals the potential threat that unresponsive application traffic presents to coexisting Internet services and QoS efforts, and introduces the requirement for a balance between application QoS and fairness. This thesis proposes a novel architecture, the Congestion Aware Packet Scheduler (CAPS), which manages and controls traffic at the point of service aggregation, in order to optimise the overall QoS of the user experience. The CAPS architecture, in contrast to traditional QoS alternatives, places no predetermined precedence on a specific traffic; instead, it adapts QoS policies to each individual’s Internet traffic profile and dynamically controls the ratio of user services to maintain an optimised QoS experience. The rationale behind this approach was to enable a QoS optimised experience to each Internet user and not just those using preferred services. Furthermore, unresponsive bandwidth intensive applications, such as Peer-to-Peer, are managed fairly while minimising their impact on coexisting services. The CAPS architecture has been validated through extensive simulations with the topologies used replicating the complexity and scale of real-network ISP infrastructures. The results show that for a number of different user-traffic profiles, the proposed approach achieves an improved aggregate QoS for each user when compared with Best effort Internet, Traditional Diffserv and Weighted-RED configurations. Furthermore, the results demonstrate that the proposed architecture not only provides an optimised QoS to the user, irrespective of their traffic profile, but through the avoidance of static resource allocation, can adapt with the Internet user as their use of services change.France Teleco