An integrated bandwidth allocation and admission control framework for the support of heterogeneous real-time traffic in class-based IP networks

The support of real-time traffic in class-based IP networks requires the reservation of resources in all the links along the end-to-end paths through appropriate queuing and forwarding mechanisms. This resource allocation should be accompanied by appropriate admission control procedures in order to guarantee that newly admitted real-time traffic flows do not cause any violation to the Quality of Service (QoS) experienced by the already established real-time traffic flows. In this paper we initially aim to highlight certain issues with respect to the areas of bandwidth allocation and admission control for the support of real-time traffic in class-based IP networks. We investigate the implications of topological placement of both the bandwidth allocation and admission control schemes. We show that the performance of bandwidth allocation and admission control schemes depends highly on the location of the employed procedures with respect to the end-users requesting the services and the various network boundaries (access, metro, core, etc.). Based on our results we conclude that the strategies for applying these schemes should be location-aware, because the performance of bandwidth allocation and admission control at different points in a class-based IP network, and for the same traffic load, can be quite different and can deviate greatly from the expected performance. Through simulations we also try to provide a quantitative view of the aforementioned deviations. Taking the implications of this “location-awareness” into account, we subsequently present a new Measurement-based Admission Control (MBAC) scheme for real-time traffic that uses measurements of aggregate bandwidth only, without keeping the state of any per-flow information. In this scheme there is no assumption made on the nature of the traffic characteristics of the real-time traffic flows, which can be of heterogeneous nature. Through simulations we show that the admission control scheme is robust with respect to traffic heterogeneity and measurement errors. We also show that our scheme compares favorably against other admission control schemes in the literature

Design and performance evaluation of switching architectures for high-speed Internet

The motivation for this thesis is the desire to build faster and scalable routers that efficiently handle the exponential traffic growth in the Internet. The Internet forwards information through a mesh of routers and switches, which has to keep up with the increasing demands of traffic. Shared-memory based switches are known to provide the best throughput-delay performance for a given memory size. In this thesis performance of commonly used memory-sharing schemes for the shared memory switches are evaluated under balanced and unbalanced bursty traffic. The scalability of shared-memory switches has been a research issue for quite sometime. One approach is to employ multiple memory modules and use them in parallel to enhance the capacity. The two well-known architectures in this category are (i) shared-multibuffer (SMB) switch architecture invented by Yamanaka et al. of Mitsubishi Electric Corporation, Japan; and (ii) the sliding-window (SW) switch architecture invented by Dr. Kumar of UTPA, Texas, USA. In this thesis, performance of these two architectures are evaluated and compared. Furthermore, in this thesis, the SW switch architecture is extended to enable priority switching to provide differentiated Quality of Service (QoS) for different traffic classes

Quality of media traffic over Lossy internet protocol networks: Measurement and improvement.

Voice over Internet Protocol (VoIP) is an active area of research in the world of communication. The high revenue made by the telecommunication companies is a motivation to develop solutions that transmit voice over other media rather than the traditional, circuit switching network. However, while IP networks can carry data traffic very well due to their besteffort nature, they are not designed to carry real-time applications such as voice. As such several degradations can happen to the speech signal before it reaches its destination. Therefore, it is important for legal, commercial, and technical reasons to measure the quality of VoIP applications accurately and non-intrusively. Several methods were proposed to measure the speech quality: some of these methods are subjective, others are intrusive-based while others are non-intrusive. One of the non-intrusive methods for measuring the speech quality is the E-model standardised by the International Telecommunication Union-Telecommunication Standardisation Sector (ITU-T). Although the E-model is a non-intrusive method for measuring the speech quality, but it depends on the time-consuming, expensive and hard to conduct subjective tests to calibrate its parameters, consequently it is applicable to a limited number of conditions and speech coders. Also, it is less accurate than the intrusive methods such as Perceptual Evaluation of Speech Quality (PESQ) because it does not consider the contents of the received signal. In this thesis an approach to extend the E-model based on PESQ is proposed. Using this method the E-model can be extended to new network conditions and applied to new speech coders without the need for the subjective tests. The modified E-model calibrated using PESQ is compared with the E-model calibrated using i ii subjective tests to prove its effectiveness. During the above extension the relation between quality estimation using the E-model and PESQ is investigated and a correction formula is proposed to correct the deviation in speech quality estimation. Another extension to the E-model to improve its accuracy in comparison with the PESQ looks into the content of the degraded signal and classifies packet loss into either Voiced or Unvoiced based on the received surrounding packets. The accuracy of the proposed method is evaluated by comparing the estimation of the new method that takes packet class into consideration with the measurement provided by PESQ as a more accurate, intrusive method for measuring the speech quality. The above two extensions for quality estimation of the E-model are combined to offer a method for estimating the quality of VoIP applications accurately, nonintrusively without the need for the time-consuming, expensive, and hard to conduct subjective tests. Finally, the applicability of the E-model or the modified E-model in measuring the quality of services in Service Oriented Computing (SOC) is illustrated