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

Providing guaranteed QoS in the hose-modeled VPN

With the development of the Internet, Internet service providers (ISPs) are required to offer revenue-generating and value-added services instead of only providing bandwidth and access services. Virtual Private Network (VPN) is one of the most important value-added services for ISPs. The classical VPN service is provided by implementing layer 2 technologies, either Frame Relay (FR) or Asynchronous Transfer Mode (ATM). With FR or ATM, virtual circuits are created before data delivery. Since the bandwidth and buffers are reserved, the QoS requirements can be naturally guaranteed. In the past few years, layer 3 VPN technologies are widely deployed due to the desirable performance in terms of flexibility, scalability and simplicity. Layer 3 VPNs are built upon IP tunnels, e.g., by using PPTP, L2TP or IPSec. Since IP is best-of-effort in nature, the QoS requirement cannot be guaranteed in layer 3 VPNs. Actually, layer 3 VPN service can only provide secure connectivity, i.e., protecting and authenticating IP packets between gateways or hosts in a VPN. Without doubt, with more applications on voice, audio and video being used in the Internet, the provision of QoS is one of the most important parts of the emerging services provided by ISPs. An intriguing question is: Is it possible to obtain the best of both layer 2 and 3 VPN? Is it possible to provide guaranteed or predictable QoS, as in layer 2 VPNs, while maintaining the flexibility and simplicity in layer 3 VPN? This question is the starting point of this study. The recently proposed hose model for VPN possesses desirable properties in terms of flexibility, scalability and multiplexing gain. However, the classic fair bandwidth allocation schemes and weighted fair queuing schemes raise the issue of low overall utilization in this model. A new fluid model for provider-provisioned virtual private network (PPVPN) is proposed in this dissertation. Based on the proposed model, an idealized fluid bandwidth allocation scheme is developed. This scheme is proven, analytically, to have the following properties: 1) maximize the overall throughput of the VPN without compromising fairness; 2) provide a mechanism that enables the VPN customers to allocate the bandwidth according to their requirements by assigning different weights to different hose flows, and thus obtain the predictable QoS performance; and 3) improve the overall throughput of the ISPs\u27 network. To approximate the idealized fluid scheme in the real world, the 2-dimensional deficit round robin (2-D DRR and 2-D DRR+) schemes are proposed. The integration of the proposed schemes with the best-effort traffic within the framework of virtual-router-based VPN is also investigated. The 2-D DRR and 2-D DER-+ schemes can be extended to multi-dimensional schemes to be employed in those applications which require a hierarchical scheduling architecture. To enhance the scalability, a more scalable non-per-flow-based scheme for output queued switches is developed as well, and the integration of this scheme within the framework of the MPLS VPN and applications for multicasting traffics is discussed. The performance and properties of these schemes are analyzed

Deadline-ordered parallel iterative matching with QoS guarantee.

by Lui Hung Ngai.Thesis (M.Phil.)--Chinese University of Hong Kong, 2000.Includes bibliographical references (leaves 56-[59]).Abstracts in English and Chinese.Chapter 1 --- Introduction --- p.1Chapter 1.1 --- Thesis Overview --- p.3Chapter 2 --- Background & Related work --- p.4Chapter 2.1 --- Scheduling problem in ATM switch --- p.4Chapter 2.2 --- Traffic Scheduling in output-buffered switch --- p.5Chapter 2.3 --- Traffic Scheduling in Input buffered Switch --- p.16Chapter 3 --- Deadline-ordered Parallel Iterative Matching (DLPIM) --- p.22Chapter 3.1 --- Introduction --- p.22Chapter 3.2 --- Switch model --- p.23Chapter 3.3 --- Deadline-ordered Parallel Iterative Matching (DLPIM) --- p.24Chapter 3.3.1 --- Motivation --- p.24Chapter 3.3.2 --- Algorithm --- p.26Chapter 3.3.3 --- An example of DLPIM --- p.28Chapter 3.4 --- Simulation --- p.30Chapter 4 --- DLPIM with static scheduling algorithm --- p.41Chapter 4.1 --- Introduction --- p.41Chapter 4.2 --- Static scheduling algorithm --- p.42Chapter 4.3 --- DLPIM with static scheduling algorithm --- p.48Chapter 4.4 --- An example of DLPIM with static scheduling algorithm --- p.50Chapter 5 --- Conclusion --- p.54Bibliography --- p.5

Congestion Control by Bandwidth-Delay Tradeoff in Very High-Speed Networks: The Case of Window-Based Control

Increasing bandwidth-delay product of high-speed wide-area networks is well-known to make conventional dynamic traffic control schemes sluggish . Still, most existing schemes employ dynamic control, among which TCP and ATM Forum\u27s rate-based flow control are prominent examples. So far, little has been investigated as to how the existing schemes will scale as bandwidth further increases up to gigabit speed and beyond. Our investigation in this paper is the first to show that dynamic control has a severe scalability problem with bandwidth increase, and to propose an entirely new approach to traffic control that overcomes the scalability problem. The essence of our approach is in exercising control in bandwidth domain rather than time domain, in order to avoid time delay in control. This requires more bandwidth than the timed counterpart, but achieves a much faster control. Furthermore, the bandwidth requirement is not excessively large because the bandwidth for smaller control delay and we call our approach Bandwidth-Latency Tradeoff (BLT). While the control in existing schemes are bound to delay, BLT is bound to bandwidth. As a fallout, BLT scales tied to bandwidth increase, rather than increasingly deteriorate as conventional schemes. Surprisingly, our approach begins to pay off much earlier than expected, even from a point where bandwidth-delay product is not so large. For instance, in a roughly AURORA-sized network, BLT far outperforms TCP on a shared 150Mbps link, where the bandwidth-delay product is around 60KB. In the other extreme where bandwidth-delay product is large, BLT outperforms TCP by as much as twenty times in terms of network power in a gigabit nationwide network. More importantly, BLT is designed to continue to scale with bandwidth increase and the performance gap is expected to widen further

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

Deadline-ordered burst-based parallel scheduling strategy for IP-over-ATM with QoS support.

Siu Chun.Thesis (M.Phil.)--Chinese University of Hong Kong, 2001.Includes bibliographical references (leaves 66-68).Abstracts in English and Chinese.Chapter 1 --- Introduction --- p.1Chapter 1.1 --- Thesis Overview --- p.3Chapter 2 --- Background and Related work --- p.4Chapter 2.1 --- Emergence of IP-over-ATM --- p.4Chapter 2.2 --- ATM architecture --- p.5Chapter 2.3 --- Scheduling issues in output-queued switch --- p.6Chapter 2.4 --- Scheduling issues in input-queued switch --- p.18Chapter 3 --- The Deadline-ordered Burst-based Parallel Scheduling Strategy --- p.23Chapter 3.1 --- Introduction --- p.23Chapter 3.2 --- Switch and queueing model --- p.24Chapter 3.2.1 --- Switch model --- p.24Chapter 3.2.2 --- Queueing model --- p.25Chapter 3.3 --- The DBPS Strategy --- p.26Chapter 3.3.1 --- Motivation --- p.26Chapter 3.3.2 --- Strategy --- p.31Chapter 3.4 --- The Deadline-ordered Burst-based Parallel Iterative Matching --- p.33Chapter 3.4.1 --- Algorithm --- p.34Chapter 3.4.2 --- An example of DBPIM --- p.35Chapter 3.5 --- Simulation results --- p.33Chapter 3.6 --- Discussions --- p.46Chapter 3.7 --- Future work --- p.47Chapter 4 --- The Quasi-static DBPIM Algorithm --- p.50Chapter 4.1 --- Introduction --- p.50Chapter 4.2 --- Quasi-static path scheduling principle --- p.51Chapter 4.3 --- Quasi-static DBPIM algorithm --- p.56Chapter 4.4 --- An example of Quasi-static DBPIM --- p.59Chapter 5 --- Conclusion --- p.63Bibliography --- p.6

Delay-oriented active queue management in TCP/IP networks

PhDInternet-based applications and services are pervading everyday life. Moreover, the growing popularity of real-time, time-critical and mission-critical applications set new challenges to the Internet community. The requirement for reducing response time, and therefore latency control is increasingly emphasized. This thesis seeks to reduce queueing delay through active queue management. While mathematical studies and research simulations reveal that complex trade-off relationships exist among performance indices such as throughput, packet loss ratio and delay, etc., this thesis intends to find an improved active queue management algorithm which emphasizes delay control without trading much on other performance indices such as throughput and packet loss ratio. The thesis observes that in TCP/IP network, packet loss ratio is a major reflection of congestion severity or load. With a properly functioning active queue management algorithm, traffic load will in general push the feedback system to an equilibrium point in terms of packet loss ratio and throughput. On the other hand, queue length is a determinant factor on system delay performance while has only a slight influence on the equilibrium. This observation suggests the possibility of reducing delay while maintaining throughput and packet loss ratio relatively unchanged. The thesis also observes that queue length fluctuation is a reflection of both load changes and natural fluctuation in arriving bit rate. Monitoring queue length fluctuation alone cannot distinguish the difference and identify congestion status; and yet identifying this difference is crucial in finding out situations where average queue size and hence queueing delay can be properly controlled and reasonably reduced. However, many existing active queue management algorithms only monitor queue length, and their control policies are solely based on this measurement. In our studies, our novel finding is that the arriving bit rate distribution of all sources contains information which can be a better indication of congestion status and has a correlation with traffic burstiness. And this thesis develops a simple and scalable way to measure its two most important characteristics, namely the mean ii and the variance of the arriving rate distribution. The measuring mechanism is based on a Zombie List mechanism originally proposed and deployed in Stabilized RED to estimate the number of flows and identify misbehaving flows. This thesis modifies the original zombie list measuring mechanism, makes it capable of measuring additional variables. Based on these additional measurements, this thesis proposes a novel modification to the RED algorithm. It utilizes a robust adaptive mechanism to ensure that the system reaches proper equilibrium operating points in terms of packet loss ratio and queueing delay under various loads. Furthermore, it identifies different congestion status where traffic is less bursty and adapts RED parameters in order to reduce average queue size and hence queueing delay accordingly. Using ns-2 simulation platform, this thesis runs simulations of a single bottleneck link scenario which represents an important and popular application scenario such as home access network or SoHo. Simulation results indicate that there are complex trade-off relationships among throughput, packet loss ratio and delay; and in these relationships delay can be substantially reduced whereas trade-offs on throughput and packet loss ratio are negligible. Simulation results show that our proposed active queue management algorithm can identify circumstances where traffic is less bursty and actively reduce queueing delay with hardly noticeable sacrifice on throughput and packet loss ratio performances. In conclusion, our novel approach enables the application of adaptive techniques to more RED parameters including those affecting queue occupancy and hence queueing delay. The new modification to RED algorithm is a scalable approach and does not introduce additional protocol overhead. In general it brings the benefit of substantially reduced delay at the cost of limited processing overhead and negligible degradation in throughput and packet loss ratio. However, our new algorithm is only tested on responsive flows and a single bottleneck scenario. Its effectiveness on a combination of responsive and non-responsive flows as well as in more complicated network topology scenarios is left for future work