Pubs:2001

<bibtex> @inproceedings{267, author={Mos,A. ; Murphy,J.}, title={Performance Monitoring of JAVA Component-Oriented Distributed Applications}, booktitle={IEEE 9th International Conference on Software,Telecommunication & Computer Networks}, year={2001}, abstract={We present a framework for monitoring the performance of component oriented distributed applications based on the Enterprise Java Beans specification. The environment leverages EJB architecture to monitor existing applications in real-time and to provide detailed run-time information that help identify performance hotspots at an object-oriented level. It is non-intrusive, portable across all EJB compliant application servers and easily extendable to accommodate new data-acquisition or graphical presentation components. Current status of the work serves as proof of concept and a complete implementation is under development.}, keywords={performance, monitoring, EJB, component-oriented, distributed systems}, pdf={performance monitoring.pdf}, } </bibtex> <bibtex> @inproceedings{268, author={Hava-Muntean,C.;McManis,J.;Murphy,J.}, title={The Influence of Web Page Images on the Performance of Web Servers}, booktitle={International Confrence on Networking}, year={2001}, abstract={In recent years World Wide Web traffic has shown phenomenal growth. The main causes are the continuing increase in the number of people navigating the Internet and the creation of millions of new Web sites. In addition, the structure of Web pages has become more complex, including not only HTML files but also other components. This has affected both the download times of Web pages and the network bandwidth required. The goal of our research is to monitor the download times of Web pages from different Web sites, and to find out to what extent the images contained in these Web pages influence these times. We also suggest some possible ways of decreasing the bandwidth requirements and download times of complex Web pages.}, keywords={Web Servers}, pdf={influence of web page.pdf}, } </bibtex> <bibtex> @inproceedings{269, author={Mos,A.;Murphy,J.}, title={New Methods for Performance Monitoring of J2EE Applications Servers}, booktitle={IEEE 8th International Conference on Telecommunications}, year={2001}, abstract={There is a growing need for high performance enterprise distributed systems that provide the scalability and availability required by modern enterprise portals and ecommerce systems. New technologies such as Enterprise Java Beans help building these systems by providing the framework to support the increasingly complex applications. Their performance, however, is not guaranteed by the technology itself and it is mostly the responsibility of the developers to build the application so that it meets the required performance needs. We present a number of approaches for monitoring existing Enterprise Java Beans applications in order to help the developers identify performance problems at an object-oriented level.}, keywords={Applications Servers}, pdf={new methods.pdf}, } </bibtex> <bibtex> @inproceedings{270, author={Hava-Muntean,C.;McManis,J.; Murphy,J.}, title={A New Dynamic Web Server}, booktitle={8th International Conference on Telecommunication}, year={2001}, abstract={The growth of traffic on the Internet and the explosion in the number of Web sites created in recent years make Web server performance an important issue for Web site designers. To improve the server's performance, it is necessary to determine the main factors affecting it before proposing new solutions for Web server design. Here we first present some experimental results on factors which influence Web server performance. We show how the number of concurrent clients accessing the server and the overall network traffic dynamics affect the performance. The details of a Web page’s composition are also studied to determine their effect on performance. Then we describe a new approach for developing a Web server, in which the server takes its performance as the clients see it into account and dynamically generates the requested Web pages. Their content depends on the traffic conditions and client capabilities. Some results are described to show the feasibility of our proposed design.}, keywords={Web Servers}, pdf={new dynamic.pdf}, } </bibtex> <bibtex> @inproceedings{271, author={Nowicki,E.;Murphy,J.}, title={Resource Allocation for Interactive traffic classes over GPRS Network}, booktitle={IEE 17th UK Teletraffic Symposium}, year={2001}, abstract={The General Packet Radio Service (GPRS) is a new bearer service for GSM that greatly simplify wireless access to packet data networks, e.g., to the Internet, to the corporate LAN or to the mobile portals. It applies a packet radio standard to transfer user data packets in a well-organized way between Mobile Stations (MS) and external packet data networks. The introduction of guaranteed performance services in GPRS networks requires detailed studies of the resource allocation and service integration issue. This paper proposes some different schemes of allocating the physical channels to mobile stations. We consider the integration of voice and data over Time Division Multiple Access wireless cellular networks. We describe different radio resource allocation algorithms, and describe their similarities and differences in the context of GSM and GPRS networks. However these algorithms can be used for diversified types of wireless networks. In this paper we consider the interactive best effort traffic class as one of the most important traffic classes in GPRS.}, keywords={GPRS }, pdf={rsource allocation.pdf}, } </bibtex> <bibtex> @inproceedings{272, author={Hava-Muntean,C.;McManis,J.; Murphy,J.;Murphy,L.}, title={A Client-Oriented Dynamic Web Server}, booktitle={IEE 17th UK Teletraffic Symposium}, year={2001}, abstract={The cost of computer systems has decreased continuously in recent years, leading to an exponential growth in the number of computer users. In such an environment, more and more Web servers have been created offering many types of information. As a result Internet traffic has grown significantly, affecting the quality of the services offered by the Web servers. We propose a new approach for designing Web servers, which takes into account client requirements and constraints, and whose implementation is based on Java servlet and applet technology. This client-orientated Web server classifies each client into one of a number of pre-defined categories. The Web page generated for a client then depends on the client's current category. A Web page generated in this way may differ from one generated for another client in its content, number of images, graphic design and structure.}, keywords={Web Servers}, pdf={Client Oriented.pdf}, } </bibtex> <bibtex> @inproceedings{350, author={M.Narbutt;L.Murphy}, title={Adaptive Playout Buffering for H.323 Voice over IP applications}, booktitle={Irish Signals & Systems Conference 2001}, year={2001}, abstract={In this paper we investigate the performance of various buffer algorithms that might be implemented in H.323 VoIP applications. The main objective of those algorithms is to minimize effect of the delay jitter. We have tested those algorithms in the Internet using H.323 VoIP terminals. Our results show that the algorithm proposed by us can achieve the lowest rate of lost packets while adding acceptably small delays.}, keywords={VoIP,Internet}, pdf={adaptive playout.pdf}, } </bibtex> <bibtex> @inproceedings{351, author={N.Cranley;L.Murphy}, title={Adaptive Quality of Service for Streamed MPEG-4 over the Internet}, booktitle={International Conference on Communications}, year={2001}, abstract={Currently multimedia is either downloaded before viewing, or streamed over a network. However, streaming real-time or near real-time applications with a specified Quality of Service (QoS) over the Internet is not yet a solved problem. The Real-time Transport Protocol (RTP) can be used to facilitate streaming, but also has the potential to support QoS. By gathering network statistics during the session and defining different QoS levels, we propose to adapt multimedia streaming to a fluctuating network load and/or client requests, thereby providing adaptive QoS. We describe the simple server and client applications we have implemented to illustrate this adaptation process.}, keywords={QoS,RTS,Internet}, pdf={adaptive quality of service.pdf}, } </bibtex> <bibtex> @inproceedings{352, author={G.Muntean;L.Murphy}, title={A Novel Feedback Controlled Multimedia Transmission Scheme}, booktitle={International Conference on Telecommunications}, year={2001}, abstract={The number of multimedia transmissions over the existing network infrastructure is continually increasing. This leads to longer periods of network congestion which affects these transmissions and hence their playout. We propose a novel feedback controlled multimedia transmission scheme in order to ensure continuous stream delivery and play-out, even in the case of network congestion. Data transmission and the exchange of control information are done via a doublechannel (TCP and UDP) link. A special protocol (Client Initiated Protocol) has been defined to provide the transmission mechanism with a reduced overhead. We describe both its components, the Client Initiated Streaming Protocol (CISP) used for control and the Client Initiated Transport Protocol (CITP) used for data transmissions. We also present the feedback scheme and describe the server's possible state transitions. A multicast approach is explored and its advantages and disadvantages outlined. We present some experimental results to show the functionality of our scheme.}, keywords={Multimedia,TCP/UDP}, pdf={novel feedback.pdf}, } </bibtex> <bibtex> @inproceedings{353, author={C.Hava;J.McManis;J.Murphy;L.Murphy}, title={A New Dynamic Web Server }, booktitle={International Conference on Telecommunications}, year={2001}, abstract={The growth of traffic on the Internet and the explosion in the number of Web sites created in recent years make Web server performance an important issue for Web site designers. To improve the server's performance, it is necessary to determine the main factors affecting it before proposing new solutions for Web server design. Here we first present some experimental results on factors which influence Web server performance. We show how the number of concurrent clients accessing the server and the overall network traffic dynamics affect the performance. The details of a Web page’s composition are also studied to determine their effect on performance. Then we describe a new approach for developing a Web server, in which the server takes its performance as the clients see it into account and dynamically generates the requested Web pages. Their content depends on the traffic conditions and client capabilities. Some results are described to show the feasibility of our proposed design.}, keywords={Internet,QoS,Web Server}, } </bibtex> <bibtex> @inproceedings{354, author={G.Muntean;L.Murphy}, title={Experimental Results for a Feedback-Controlled Multimedia Transmission System}, booktitle={17th IEE UK Teletraffic Symposium}, year={2001}, abstract={Multimedia transmissions over IP networks have increased significantly in recent years. Both user expectations and traffic levels have increased as well. One way to take these into account is for senders to receive and respond to feedback about the quality of their transmissions. We have built a feedback controlled multimedia transmission system, and we have tested and analysed the influence of various factors on transmission performance. This paper presents the results of some of these tests, and suggests some solutions for improving the quality of multimedia transmissions under network congestion.}, keywords={Multimedia,IP}, pdf={experimental.pdf}, } </bibtex> <bibtex> @inproceedings{355, author={M.Narbutt;L.Murphy}, title={Adaptive Playout Buffering for Audio/Video transmission over the Internet}, booktitle={17th IEE UK Teletraffic Symposium}, year={2001}, abstract={Transmitting real-time audio/video over the Internet is very difficult due to packet loss and jitter. These parameters vary depending on the locations of the senders and receivers, with typical packet loss rates of 0?20% and one-way delays of 5-500 ms. Delay variations occur within and across audio and video streams, complicating the synchronization process. One possibility for reducing jitter involves buffering audio and video packets at the receiver, so that slower packets arrive in time to be played out in the correct sequence at the appropriate times. This paper presents various adaptive playout buffer algorithms that minimize the effect of delay jitter. We evaluate their effectiveness through experiments based on a real network and compare their performance in terms of delay/packet loss ratios. Although the main focus of this paper is the playout buffering for audio, the synchronization between audio and video streams is also specified.Transmitting real-time audio/video over the Internet is very difficult due to packet loss and jitter. These parameters vary depending on the locations of the senders and receivers, with typical packet loss rates of 0?20% and one-way delays of 5-500 ms. Delay variations occur within and across audio and video streams, complicating the synchronization process. One possibility for reducing jitter involves buffering audio and video packets at the receiver, so that slower packets arrive in time to be played out in the correct sequence at the appropriate times. This paper presents various adaptive playout buffer algorithms that minimize the effect of delay jitter. We evaluate their effectiveness through experiments based on a real network and compare their performance in terms of delay/packet loss ratios. Although the main focus of this paper is the playout buffering for audio, the synchronization between audio and video streams is also specified.}, keywords={Internet,Online-Streaming}, pdf={adaptive buffering.pdf}, } </bibtex> <bibtex> @inproceedings{356, author={N.Cranley;L.Murphy}, title={Adaptive Quality of Service for Streamed MPEG-4 over the Internet}, booktitle={17th IEE UK Teletaffic Symposium}, year={2001}, abstract={Currently multimedia is either downloaded before viewing, or streamed over a network. However, streaming real-time or near real-time applications with a specified Quality of Service (QoS) over the Internet is not yet a solved problem. The Real-time Transport Protocol (RTP) can be used to facilitate streaming, but also has the potential to support QoS. By gathering network statistics during the session and defining different QoS levels, we propose to adapt multimedia streaming to a fluctuating network load and/or client requests, thereby providing adaptive QoS. We describe the simple server and client applications we have implemented to illustrate this adaptation process.}, keywords={RTP,QoS}, pdf={adaptive quality.pdf}, } </bibtex> <bibtex> @inproceedings{465, author={H.Melvin;L.Murphy}, title={An Investigation into the use of Synchronised Time to Improve VoIP Service }, booktitle={IEI/IEE Symposium on Telecommunications Systems Research }, year={2001}, abstract={This paper presents an overview of work in progress relating to the use of synchronised time in Voice over IP (VoIP) networks. One of the principal limita- tions of conventional IP networks is the best-e�ort service that they provide. In recent years, the In- ternet Engineering Task Force (IETF) have created the multimedia data and control architecture which incorporates protocols such as the Realtime Trans- port Protocol (RTP), RTP Control Protocol (RTCP) and Real Time Streaming Protocol (RTSP). Whilst these protocols assist in delivery of multimedia traÆc, the lack of bounds on end-to-end delay still presents signi�cant problems for interactive applications such as VoIP. The ITU-T recommendation G.114 speci�es that a round-trip-delay of 300ms should not be ex- ceeded and the Plain Old Telephone System (POTS) generally does much better than this. Along with RTP and RTCP, various approaches have been pro- posed and taken to improve the performance of VoIP networks from an interactive viewpoint. These range from sender-based codec measures to network-based policies that di�erentiate between traÆc ows and �- nally to various bu�er schemes at the receiver that balance overall delay with receiver packet loss due to late arrival. This paper summarises much of this work but focuses on receiver-based measures. RTP timestamps are principally used to enable re- ceiver hosts to monitor and react to the inter-packet delay variance (jitter). RTCP packet timestamps fur- ther enable sender hosts to periodically determine round-trip-times (RTT). Such calculations do not re- quire that end host clocks are synchronised. In this paper, a scheme is described whereby syn- chronised time is available to hosts in a VoIP ses- sion. As Global Positioning System (GPS) receivers become increasingly cost-e�ective and always-on In- ternet connections more prevalent, VoIP hosts can avail of synchronised time either via GPS or Network Time Protocol (NTP). This time synchronisation en- ables each host to know precisely the end-to-end de- lays on a packet-by-packet basis. This is a signi�- cant improvement on the present situation whereby sender hosts can periodically determine round-trip- times and thus estimate one-way-delays. Most adap- tive receiver bu�er schemes do not consider such es- timates and are designed to minimise delay at the expense of tolerable packet loss due to late arrival and some distortion of inter-talkspurt silence peri- ods. In many instances, the performance of private IP networks and the localised public Internet will op- erate well within the bounds of G.114. In such sit- uations, a �xed end-to-end delay within the G.114 bounds yet marginally above the actual end-to-end performance would result in a tolerable delay with no late packet loss and no distortion of silence pe- riods. With synchronised time, supporting protocols and receiver-based intelligence, such a system can be implemented.}, keywords={VoIP,IP Networks}, pdf={an investigation.pdf}, } </bibtex> <bibtex> @inproceedings{466, author={M.Narbutt;L.Murphy}, title={Adaptive Anti-Jitter Mechanism for Multi-Party Conferencing in a H.323 Multi Point Control Unit}, booktitle={IEI/IEE Symposium on Tlecommunications System Research}, year={2001}, abstract={In this paper we propose a mechanism that can support multi-party conferencing. The main objective of this mechanism is to minimize effect of delay jitter. We have tested this mechanism in the Internet using H.323 terminals and H.323 Multipoint Control Unit. Our results show that the playout algorithms traditionally designed to work in the receiving endpoints can also be successfully implemented in the Multipoint Control Unit. As a result one can lower the rate of lost packets due to their late arrival.}, keywords={Delay,Jitter}, pdf={adaptive anti jitter.pdf}, } </bibtex>