Quality-of-service management in IP networks

Quality of Service (QoS) in Internet Protocol (IF) Networks has been the subject of active research over the past two decades. Integrated Services (IntServ) and Differentiated Services (DiffServ) QoS architectures have emerged as proposed standards for resource allocation in IF Networks. These two QoS architectures support the need for multiple traffic queuing systems to allow for resource partitioning for heterogeneous applications making use of the networks. There have been a number of specifications or proposals for the number of traffic queuing classes (Class of Service (CoS)) that will support integrated services in IF Networks, but none has provided verification in the form of analytical or empirical investigation to prove that its specification or proposal will be optimum. Despite the existence of the two standard QoS architectures and the large volume of research work that has been carried out on IF QoS, its deployment still remains elusive in the Internet. This is not unconnected with the complexities associated with some aspects of the standard QoS architectures. [Continues.

A methodological approach to BISDN signalling performance

Sophisticated signalling protocols are required to properly handle the complex multimedia, multiparty services supported by the forthcoming BISDN. The implementation feasibility of these protocols should be evaluated during their design phase, so that possible performance bottlenecks are identified and removed. In this paper we present a methodology for evaluating the performance of BISDN signalling systems under design. New performance parameters are introduced and their network-dependent values are extracted through a message flow model which has the capability to describe the impact of call and bearer control separation on the signalling performance. Signalling protocols are modelled through a modular decomposition of the seven OSI layers including the service user to three submodels. The workload model is user descriptive in the sense that it does not approximate the direct input traffic required for evaluating the performance of a layer protocol; instead, through a multi-level approach, it describes the actual implications of user signalling activity for the general signalling traffic. The signalling protocol model is derived from the global functional model of the signalling protocols and information flows using a network of queues incorporating synchronization and dependency functions. The same queueing approach is followed for the signalling transfer network which is used to define processing speed and signalling bandwidth requirements and to identify possible performance bottlenecks stemming from the realization of the related protocols

An Adaptive Scheme for Admission Control in ATM Networks

This paper presents a real time front-end admission control scheme for ATM networks. A call management scheme which uses the burstiness associated with traffic sources in a heterogeneous ATM environment to effect dynamic assignment of bandwidth is presented. In the proposed scheme, call acceptance is based on an on-line evaluation of the upper bound on cell loss probability which is derived from the estimated distribution of the number of calls arriving. Using this scheme, the negotiated quality of service will be assured when there is no estimation error. The control mechanism is effective when the number of calls is large, and tolerates loose bandwidth enforcement and loose policing control. The proposed approach is very effective in the connection oriented transport of ATM networks where the decision to admit new traffic is based on thea priori knowledge of the state of the route taken by the traffic

Hybrid switching : converging packet and TDM flows in a single platform

Optical fibers have brought fast and reliable data transmission to today’s network. The immense fiber build-out over the last few years has generated a wide array of new access technologies, transport and network protocols, and next-generation services in the Local Area Network (LAN), Metropolitan Area Network (MAN), and Wide Area Network (WAN). All these different technologies, protocols, and services were introduced to address particular telecommunication needs. To remain competitive in the market, the service providers must offer most of these services, while maintaining their own profitability. However, offering a large variety of equipment, protocols, and services posses a big challenge for service carriers because it requires a huge investment in different technology platforms, lots of training of staff, and the management of all these networks. In today’s network, service providers use SONET (Synchronous Optical NETwork) as a basic TDM (Time Division Multiplexing) transport network. SONET was primarily designed to carry voice traffic from telephone networks. However, with the explosion of traffic in the Internet, the same SONET based TDM network is optimized to support increasing demand for packet based Internet network services (data, voice, video, teleconference etc.) at access networks and LANs. Therefore the service providers need to support their Internet Protocol (IP) infrastructure as well as in the legacy telephony infrastructure. Supporting both TDM and packet services in the present condition needs multilayer operations which is complex, expensive, and difficult to manage. A hybrid switch is a novel architecture that combines packets (IP) and TDM switching in a unified access platform and provides seamless integration of access networks and LANs with MAN/WAN networks. The ability to fully integrate these two capabilities in a single chassis will allow service providers to deploy a more cost effective and flexible architecture that can support a variety of different services. This thesis develops a hybrid switch which is capable of offering bundled services for TDM switching and packet routing. This is done by dividing the switch’s bandwidth into VT1.5 (Virtual Tributary -1.5) channels and providing SONET based signaling for routing the data and controlling the switch’s resources. The switch is a TDM based architecture which allows each switch’s port to be independently configured for any mixture of packet and TDM traffic, including 100% packet and 100% TDM. This switch allows service providers to simplify their edge networks by consolidating the number of separate boxes needed to provide fast and reliable access. This switch also reduces the number of network management systems needed, and decreases the resources needed to install, provision and maintain the network because of its ability to “collapse” two network layers into one platform. The scope of this thesis includes system architecture, logic implementation, and verification testing, and performance evaluation of the hybrid switch. The architecture consists of ingress/egress ports, an arbiter and a crossbar. Data from ingress ports is carried to the egress ports via VT1.5 channels which are switched at the cross point of the crossbar. The crossbar setup and channel assignments at ingress port are done by the arbiter. The design was tested by simulation and the hardware cost was estimated. The performance results showed that the switch is non-blocking, provide differentiated service, and has an overall effective throughput of 80%. This result is a significant step towards the goal of building a switch that can support multiprotocol and provide different network capabilities into one platform. The long-term goal of this project is to develop a prototype of the hybrid switch with broadband capability

Distributed multimedia systems

A distributed multimedia system (DMS) is an integrated communication, computing, and information system that enables the processing, management, delivery, and presentation of synchronized multimedia information with quality-of-service guarantees. Multimedia information may include discrete media data, such as text, data, and images, and continuous media data, such as video and audio. Such a system enhances human communications by exploiting both visual and aural senses and provides the ultimate flexibility in work and entertainment, allowing one to collaborate with remote participants, view movies on demand, access on-line digital libraries from the desktop, and so forth. In this paper, we present a technical survey of a DMS. We give an overview of distributed multimedia systems, examine the fundamental concept of digital media, identify the applications, and survey the important enabling technologies.published_or_final_versio

Does it work? evaluating a new pay system

This report focuses on the evaluation of the impact of new pay systems in large, unionised multi-site organisations by the organisations themselves. Evaluation of the effectiveness of a pay system, however, does not take place in a vacuum and relates to the aims and objectives of the pay system concerned. Moreover, evaluation is not an end in itself. It is, therefore, relevant to consider if any further steps were taken as a result of evaluation. Accordingly our research questions were: • What were the aims and objectives of organisations when introducing new pay arrangements? • What data did organisations collect and review to inform their evaluation? • What steps have organisations taken as a result of their evaluation? We re-appraised our data from 10 NHS trusts in England which had introduced some innovations in pay and grading in the 1990s. Additionally, we looked at seven multi-site unionised organisations outside the NHS in both the public and private sectors, which had recently made changes to their reward systems, carrying out interviews and inspecting documents. The main output is a template for the evaluation of Agenda for Change by NHS organisations

Multiple description image and video coding for P2P transmissions

Peer-to-Peer (P2P) media streaming is, nowadays, a very attractive topic due to the bandwidth available to serve demanding content scales. A key challenge, however, is making content distribution robust to peer transience. Multiple description coding (MDC) has, indeed, proven to be very effective with problems concerning the packets’ losses, since it generates several descriptions and may reconstruct the original information with any number of descriptions that may reach the decoder. Therefore multiple descriptions may be effective for robust peer-to-peer media streaming. In this dissertation, it will not only be showed that, but also that varying the redundancy level of description on the fly may lead to a better performance than the one obtained without varying this parameter. Besides that, it is shown, as well, that varying the Bitrate on the fly outperforms the redundancy on it. Furthermore, the redundancy and the Bitrate were varied simultaneously. Thus, it is shown that this variation is more efficient when the packet loss is high. The experiments reported above were done using an experimental test bed developed for this purpose at the NMCG lab of the University of Beira Interior. It was also used the REGPROT, a video encoder developed by our research team, to splitted the video into multiple descriptions, which were, later, distributed among the peers in the test bed. After the request of the client, the referred encoder decoded the descriptions as they were being received.Fundação para a Ciência e a Tecnologia (FCT