Design of an integrated environment for adaptive multimedia document presentation through real time monitoring

The retrieval of multimedia objects is influenced by factor such as throughput and maximum delay offered by the network, and has to be carried out in accordance with the specification of object relationships. Many current network architectures address QoS from a provider' s point of view and analyze network performance, failing to comprehensively address the quality needs of applications. The work presented in this paper concerns the development of an integrated environment for creation and retrieval of multimedia documents, that intends to preserve the coherence between the different media, even when the process is confronted with a temporary lack of communication resources. This environment implements a communication system that, address QoS from the application's point of view and can help in handling variations in network resources availability through a real-time monitoring over these object relationships

On the Scaling of Feedback Algorithms for Very Large Multicast Groups

Feedback from multicast group members is vital for many multicast protocols. In order to avoid feedback implosion in very large groups feedback algorithms with well behaved scaling-properties must be chosen. In this paper we analyse the performance of three typical feedback algorithms described in the literature. Apart from the basic trade-off between feedback latency and response duplicates we especially focus on the algorithms' sensitivity to the quality of the group size estimation. Based on this analysis we give recommendations for the choice of well behaved feedback algorithms that are suitable for very large groups

Signaling for Internet Telephony

Internet telephony must offer the standard telephony services.However, the transition to Internet-based telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network(PSTN). The Session Initiation Protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems, including conferences and point-to-point calls. SIP supports unicast, mesh and multicast conferences, as well as combinations of these modes. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queueing by a small set of call handling primitives. SIP implementations can re-use parts of other Internet service protocols such as HTTP and the Real-Time Stream Protocol (RTSP). In this paper, we describe SIP, and show how its basic primitives can be used to construct a wide range of telephony services

QoS architecture and monitoring for videoconferencing applications

Mémoire numérisé par la Direction des bibliothèques de l'Université de Montréal

PERFORMANCE CHARACTERISATION OF IP NETWORKS

The initial rapid expansion of the Internet, in terms of complexity and number of hosts, was followed by an increased interest in its overall parameters and the quality the network offers. This growth has led, in the first instance, to extensive research in the area of network monitoring, in order to better understand the characteristics of the current Internet. In parallel, studies were made in the area of protocol performance modelling, aiming to estimate the performance of various Internet applications. A key goal of this research project was the analysis of current Internet traffic performance from a dual perspective: monitoring and prediction. In order to achieve this, the study has three main phases. It starts by describing the relationship between data transfer performance and network conditions, a relationship that proves to be critical when studying application performance. The next phase proposes a novel architecture of inferring network conditions and transfer parameters using captured traffic analysis. The final phase describes a novel alternative to current TCP (Transmission Control Protocol) models, which provides the relationship between network, data transfer, and client characteristics on one side, and the resulting TCP performance on the other, while accounting for the features of current Internet transfers. The proposed inference analysis method for network and transfer parameters uses online nonintrusive monitoring of captured traffic from a single point. This technique overcomes limitations of prior approaches that are typically geared towards intrusive and/or dual-point offline analysis. The method includes several novel aspects, such as TCP timestamp analysis, which allows bottleneck bandwidth inference and more accurate receiver-based parameter measurement, which are not possible using traditional acknowledgment-based inference. The the results of the traffic analysis determine the location of the eventual degradations in network conditions relative to the position of the monitoring point. The proposed monitoring framework infers the performance parameters of network paths conditions transited by the analysed traffic, subject to the position of the monitoring point, and it can be used as a starting point in pro-active network management. The TCP performance prediction model is based on the observation that current, potentially unknown, TCP implementations, as well as connection characteristics, are too complex for a mathematical model. The model proposed in this thesis uses an artificial intelligence-based analysis method to establish the relationship between the parameters that influence the evolution of the TCP transfers and the resulting performance of those transfers. Based on preliminary tests of classification and function approximation algorithms, a neural network analysis approach was preferred due to its prediction accuracy. Both the monitoring method and the prediction model are validated using a combination of traffic traces, ranging from synthetic transfers / environments, produced using a network simulator/emulator, to traces produced using a script-based, controlled client and uncontrolled traces, both using real Internet traffic. The validation tests indicate that the proposed approaches provide better accuracy in terms of inferring network conditions and predicting transfer performance in comparison with previous methods. The non-intrusive analysis of the real network traces provides comprehensive information on the current Internet characteristics, indicating low-loss, low-delay, and high-bottleneck bandwidth conditions for the majority of the studied paths. Overall, this study provides a method for inferring the characteristics of Internet paths based on traffic analysis, an efficient methodology for predicting TCP transfer performance, and a firm basis for future research in the areas of traffic analysis and performance modelling

Local Coordination for Interpersonal Communication Systems

The decomposition of complex applications into modular units is anacknowledged design principle for creating robust systems and forenabling the flexible re-use of modules in new applicationcontexts. Typically, component frameworks provide mechanisms and rulesfor developing software modules in the scope of a certain programmingparadigm or programming language and a certain computing platform. Forexample, the JavaBeans framework is a component framework for thedevelopment of component-based systems -- in the Java environment.In this thesis, we present a light-weight, platform-independentapproach that views a component-based application as a set of ratherloosely coupled parallel processes that can be distributed on multiplehosts and are coordinated through a protocol. The core of ourframework is the Message Bus (Mbus): an asynchronous, message-orientedcoordination protocol that is based on Internet technologies andprovides group communication between application components.Based on this framework, we have developed a local coordinationarchitecture for decomposed multimedia conferencing applications thatis designed for endpoint and gateway applications. One element of thisarchitecture is an Mbus-based protocol for the coordination of callcontrol components in conferencing applications

IPtel - um sistema de IPtel com suporte para vídeo utilizando o protocolo SIP

Estudos recentes apontam que o tráfego de dados em breve excederá o tráfego telefónico, se tal já não tiver acontecido. Estes indicadores, juntamente com mais e melhores acessos à Internet, tornam cada vez maior o interesse em transportar voz e vídeo sobre redes de dados. Neste contexto nasce a Telefonia sobre IP, que oferece através desta infra-estrutura a oportunidade de criar sistemas globais de comunicação multimédia. A redução de custos e a facilidade na implementação de serviços inovadores são argumentos que justificam a forte evolução da IPtel e a tendência eventual de substituir a rede telefónica analógica. O Session Initiation Protocol (SIP), utilizado no desenvolvimento deste serviço, é um protocolo de sinalização e controlo de chamadas entre dois ou mais participantes, que tem ganho uma grande aceitação por parte de empresas ligadas ao mundo das comunicações. Desenvolvido com uma arquitectura normalizada e aberta pela Internet Engineering Task Force (IETF), espera-se que o SIP tenha o mesmo impacto no mundo das comunicações IP que o SMTP teve no e-mail e o HTTP na Web. O SIP anuncia ainda a convergência dos equipamentos e serviços de comunicações, permitindo integrar facilmente serviços de Internet como Web, e-mail, correio de voz, mensagens instantâneas, colaboração multimédia e presença (informação sobre se um utilizador está ou não disponível para comunicar). Nesta dissertação é feito um estudo sobre a evolução das diferentes partes que integram o serviço IPtel. São ainda referidas as vantagens na criação de novos serviços e obstáculos a ultrapassar por esta tecnologia de modo a poderem consolidar-se no mercado das comunicações. São apresentados diversos protocolos tipicamente usados na arquitectura protocolar da IPtel e que serão estudados, para suportar a criação do serviço sIPtel. É feita uma apresentação do serviço de Telefonia sobre IP e é explicada a arquitectura e o funcionamento do protocolo SIP, utilizado para o desenvolvimento da parte de sinalização do sIPtel. É ainda detalhado o desenvolvimento de um serviço que permite a criação, controlo e finalização de sessões de áudio e vídeo entre dois utilizadores através do protocolo SIP e por fim são realizados testes de modo a avaliar a capacidade de interoperabilidade do serviço implementado. Palavras chave: Telefonia sobre IP, Protocolos, Sinalização, Codificadores de áudio, Codificadores de vídeo, Java

Equation-Based Congestion Control for Unicast and Multicast Data Streams

We believe that the emergence of congestion control mechanisms for relatively-smooth congestion control for unicast and multicast traffic can play a key role in preventing the degradation of end-to-end congestion control in the public Internet, by providing a viable alternative for multimedia flows that would otherwise be tempted to avoid end-to-end congestion control altogether. The design of good congestion control mechanisms is a hard problem, even more so for multicast environments where scalability issues are much more of a concern than for unicast. In this dissertation, equation-based congestion control is presented as an alternative form of congestion control to the well-known TCP protocol. We focus on areas of equation-based congestion control which were not yet well understood and for which no adequate solutions existed. Starting from a unicast congestion control mechanism which in contrast to TCP provides smooth rate changes, we extend equation-based congestion control in several ways. We investigate how it can work together with applications which can only operate in a very limited region of available bandwidth and whose rate can thus not be adapted to the network conditions in the usual way. Such a congestion control mechanism can also complement conventional equation-based congestion control in regimes where available bandwidth is too low for further rate reduction. When extending unicast congestion control to multicast, it is of paramount importance to ensure that changes in the network conditions anywhere in the multicast tree are reported back to the sender as quickly as possible to allow the sender to adjust the rate accordingly. A scalable feedback mechanism that allows timely congestion feedback in the face of potentially very large receiver sets is one of the contributions of this dissertation. But also other components of a congestion control protocol, such as the rate increase/decrease policy or the slow-start mechanism, need to be adjusted to be able to use them in a multicast environment. Our resulting multicast congestion control protocol was implemented in a simulation environment for extensive protocol testing and turned into a library for the use in real-world applications. In addition, a simple video transmission tool was built for test purposes that uses this congestion control library