The QoSxLabel: a quality of service cross layer label

A quality of service cross layer label

Towards a new generation of transport services adapted to multimedia application

Une connexion d'ordre et de fiabilité partiels (POC, partial order connection) est une connexion de transport autorisée à perdre certains objets mais également à les délivrer dans un ordre éventuellement différent de celui d'émission. L'approche POC établit un lien conceptuel entre les protocoles sans connexion au mieux et les protocoles fiables avec connexion. Le concept de POC est motivé par le fait que dans les réseaux hétérogènes sans connexion tels qu'Internet, les paquets transmis sont susceptibles de se perdre et d'arriver en désordre, entraînant alors une réduction des performances des protocoles usuels. De plus, on montre qu'un protocole associé au transport d'un flux multimédia permet une réduction très sensible de l'utilisation des ressources de communication et de mémorisation ainsi qu'une diminution du temps de transit moyen. Dans cet article, une extension temporelle de POC, nommée TPOC (POC temporisé), est introduite. Elle constitue un cadre conceptuel permettant la prise en compte des exigences de qualité de service des applications multimédias réparties. Une architecture offrant un service TPOC est également introduite et évaluée dans le cadre du transport de vidéo MPEG. Il est ainsi démontré que les connexions POC comblent, non seulement le fossé conceptuel entre les protocoles sans connexion et avec connexion, mais aussi qu'ils surpassent les performances des ces derniers lorsque des données multimédias (telles que la vidéo MPEG) sont transportées

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

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

Using the Java Media Framework to build Adaptive Groupware Applications

Realtime audio and video conferencing has not yet been satisfactorily integrated into web-based groupware environments. Conferencing tools are at best only loosely linked to other parts of a shared working environment, and this is in part due to their implications for resource allocation and management. The Java Media Framework offers a promising means of redressing this situation. This paper describes an architecture for integrating the management of video and audio conferences into the resource allocation mechanism of an existing web-based groupware framework. The issue of adaptation is discussed and a means of initialising multimedia session parameters based on predicted QoS is described

Script-Based QOS Specifications for Multimedia Presentations

Multimedia presentations can convey information not only by the sequence of events but by their timing. The correctness of such presentations thus depends on the timing of events as well as their sequence and content. This paper introduces a formal specification language for playback of real-time presentations. The main contribution of this language is a quality of service (QOS) specification that relaxes resolution and synchronization requirements for playback. Our definitions give a precise meaning to the correctness of a presentation. This specification language will form the basis for a QOS interface for reservation of operating system resources

Multimedia delivery in the future internet

The term “Networked Media” implies that all kinds of media including text, image, 3D graphics, audio and video are produced, distributed, shared, managed and consumed on-line through various networks, like the Internet, Fiber, WiFi, WiMAX, GPRS, 3G and so on, in a convergent manner [1]. This white paper is the contribution of the Media Delivery Platform (MDP) cluster and aims to cover the Networked challenges of the Networked Media in the transition to the Future of the Internet. Internet has evolved and changed the way we work and live. End users of the Internet have been confronted with a bewildering range of media, services and applications and of technological innovations concerning media formats, wireless networks, terminal types and capabilities. And there is little evidence that the pace of this innovation is slowing. Today, over one billion of users access the Internet on regular basis, more than 100 million users have downloaded at least one (multi)media file and over 47 millions of them do so regularly, searching in more than 160 Exabytes1 of content. In the near future these numbers are expected to exponentially rise. It is expected that the Internet content will be increased by at least a factor of 6, rising to more than 990 Exabytes before 2012, fuelled mainly by the users themselves. Moreover, it is envisaged that in a near- to mid-term future, the Internet will provide the means to share and distribute (new) multimedia content and services with superior quality and striking flexibility, in a trusted and personalized way, improving citizens’ quality of life, working conditions, edutainment and safety. In this evolving environment, new transport protocols, new multimedia encoding schemes, cross-layer inthe network adaptation, machine-to-machine communication (including RFIDs), rich 3D content as well as community networks and the use of peer-to-peer (P2P) overlays are expected to generate new models of interaction and cooperation, and be able to support enhanced perceived quality-of-experience (PQoE) and innovative applications “on the move”, like virtual collaboration environments, personalised services/ media, virtual sport groups, on-line gaming, edutainment. In this context, the interaction with content combined with interactive/multimedia search capabilities across distributed repositories, opportunistic P2P networks and the dynamic adaptation to the characteristics of diverse mobile terminals are expected to contribute towards such a vision. Based on work that has taken place in a number of EC co-funded projects, in Framework Program 6 (FP6) and Framework Program 7 (FP7), a group of experts and technology visionaries have voluntarily contributed in this white paper aiming to describe the status, the state-of-the art, the challenges and the way ahead in the area of Content Aware media delivery platforms

Multimedia congestion control: circuit breakers for unicast RTP sessions

The Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can lead to uncontrolled packet loss and a resulting deterioration of the user's multimedia experience. The congestion control algorithm acts as a safety measure by stopping RTP flows from using excessive resources and protecting the network from overload. At the time of this writing, however, while there are several proprietary solutions, there is no standard algorithm for congestion control of interactive RTP flows. This document does not propose a congestion control algorithm. It instead defines a minimal set of RTP circuit breakers: conditions under which an RTP sender needs to stop transmitting media data to protect the network from excessive congestion. It is expected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be able to operate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by these RTP circuit breaker algorithms