An Overview of Centralised Middleware Components for Sensor Networks

Sensors are increasingly becoming part of our daily lives: motion detection, lighting control, environmental monitoring, and keeping track of energy consumption all rely on sensors. Combining data from this wide variety of sensors will result in new and innovative applications. However, access to these sensors – or the networks formed by them – is often provided via proprietary protocols and data formats, thereby obstructing the development of applications. To overcome such issues, middleware components have been employed to provide a universal interface to the sensor networks, hiding vendor-specific details from application developers. The scientific literature contains many descriptions of middleware components for sensor networks, with ideas from various fields of research. Recently, much attention in literature is aimed at what we, in this paper, define as ‘centralised’ middleware components. These components consider sensor networks that have no capacity – in terms of memory, data storage, and cpu power – to run middleware components (partially) on the sensor nodes. Often, viewed from the position of the middleware component, these sensor networks function as simple data providers for applications In this paper we introduce the term ‘centralised’ for such middleware components, guided by a literature review of existing middleware components for sensor networks. We describe their general architecture, give a description of a representative set of four centralised middleware components, and discuss advantages and disadvantages of these components. Finally, we identify directions of further research that will impact centralised middleware systems in the near future

An intelligent radio access network selection and optimisation system in heterogeneous communication environments

PhDThe overlapping of the different wireless network technologies creates heterogeneous communication environments. Future mobile communication system considers the technological and operational services of heterogeneous communication environments. Based on its packet switched core, the access to future mobile communication system will not be restricted to the mobile cellular networks but may be via other wireless or even wired technologies. Such universal access can enable service convergence, joint resource management, and adaptive quality of service. However, in order to realise the universal access, there are still many pending challenges to solve. One of them is the selection of the most appropriate radio access network. Previous work on the network selection has concentrated on serving the requesting user, but the existing users and the consumption of the network resources were not the main focus. Such network selection decision might only be able to benefit a limited number of users while the satisfaction levels of some users are compromised, and the network resources might be consumed in an ineffective way. Solutions are needed to handle the radio access network selection in a manner that both of the satisfaction levels of all users and the network resource consumption are considered. This thesis proposes an intelligent radio access network selection and optimisation system. The work in this thesis includes the proposal of an architecture for the radio access network selection and optimisation system and the creation of novel adaptive algorithms that are employed by the network selection system. The proposed algorithms solve the limitations of previous work and adaptively optimise network resource consumption and implement different policies to cope with different scenarios, network conditions, and aims of operators. Furthermore, this thesis also presents novel network resource availability evaluation models. The proposed models study the physical principles of the considered radio access network and avoid employing assumptions which are too stringent abstractions of real network scenarios. They enable the implementation of call level simulations for the comparison and evaluation of the performance of the network selection and optimisation algorithms

Enabling Layered Video Coding for IMS-Based IPTV Home Services

Nowadays IPTV services are gaining attention from both providers and end users. There is a large effort toward the integration of these services into emerging next-generation network architectures. In particular, one of the most relevant solutions is being proposed by ETSI-TISPAN and is based on the IP multimedia subsystem. This article focuses on introducing layered video coding into TISPAN IMS-based IPTV architecture, allowing cost-effective efficient solutions both for residential users and providers (e.g., flexible support of heterogeneous devices, live mosaics, adaptive video quality based on device and/or network capabilities). The advantages of using layered video coding in the TISPAN IPTV solution are analyzed and illustrated with a set of use cases. Furthermore, this solution has been integrated into a multimedia testbed in order to validate the presented proposal

Interim research assessment 2003-2005 - Computer Science

This report primarily serves as a source of information for the 2007 Interim Research Assessment Committee for Computer Science at the three technical universities in the Netherlands. The report also provides information for others interested in our research activities

Enhancement of perceived quality of service for voice over internet protocol systems

Voice over Internet Protocol (WIP) applications are becoming more and more popular in the telecommunication market. Packet switched V61P systems have many technical advantages over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible use of the bandwidth, lower cost and enhanced security. However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed in the VoIP services. In fact, most current Vol]P services can not provide as good a voice quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived speech quality as do application layer impairment factors, such as codec rate and audio features. Current perceived Quality of Service (QoS) methods are mainly designed to be used in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a challenge to measure perceived speech quality correctly in V61P system and to enhance user perceived speech quality for VoIP system. The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless systems in the context of V61P, and to develop novel and efficient methods to enhance the user perceived speech quality for emerging V61P services especially in mobile V61P environment. The main contributions of the thesis are threefold: (1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation of PESQ performance in mobile VoIP environment was undertaken and included setting up a PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto- PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was investigated and main problems causing inaccurate PESQ score (improper time-alignment in the PESQ algorithm) were discovered . Calibration issues for a safe and proper PESQ testing in mobile environment were also discussed in the thesis. (2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance. Results show that the proposed algorithm can increase user perceived quality without consuming too much processing power when tested in live wireless VbIP networks. (3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority for the beginning of a voiced segment). The results gathered on a simulation and emulation test platform shows that the combined method provides a better user perceived speech quality than separate adaptive sender bit rate or packet priority marking methods