TCP-friendly layered multicast protocol for multimedia streaming

Layered multicast protocol (LMP) is designed to support simultaneous and real-time multimedia content distribution to large number ofdisparate receivers across the heterogeneous Internet. However, the current layered multicast protocols (LMPs) use the techniques that do not correctly model TCP behaviour, which results in the protocols can't fairly compete with TCP. TCP-friendly layered multicast protocol (TjLMP) is designed to solve the problems with the current LMPs. It use the techniques that closely model TCP behaviour, i.e. scalable RTT estimation and 2-step loss event rate filtering. The techniques enable TjLMP to compete with TCP and get fair amount ofbandwidth share. TjLMP is evaluated using ns2 simulation and the results are compared with the other LMPs

Large Scale Content Distribution Protocols

This paper introduces large scale content distribution pro- tocols, which are capable of scaling to massive numbers of users and providing low delay end-to-end delivery. Delivery of files and static objects is described, with real-time con- tent streaming being outside the scope of this paper. The focus is on solutions provided by the IETF Reliable Multi- cast Transport Working Group. More precisely, the paper explains FLUTE, ALC and the associated building blocks. Then it discusses how these components are used in the Multimedia Broadcast Multicast Service (MBMS) for 3G systems and in the IP Datacast (IPDC) service for Digital Video Broadcast for Handheld devices (DVB-H)

Unicast UDP Usage Guidelines for Application Designers

Publisher PD

FLUTE - File Delivery over Unidirectional Transport

Internet Engineering Task Force (IETF) Request for Comments: 6726This document defines File Delivery over Unidirectional Transport (FLUTE), a protocol for the unidirectional delivery of files over the Internet, which is particularly suited to multicast networks. The specification builds on Asynchronous Layered Coding, the base protocol designed for massively scalable multicast distribution. This document obsoletes RFC 3926

Transport multipoint fiable à très grande échelle : intégration de critères de coût en environnement Internet hybride satellite / terrestre

Le travail effectué aborde la problématique des services de communication multipoints fiables à grande échelle. Dans ce contexte, la possibilité de déployer un tel service au moyen d'un satellite géostationnaire émettant en bande Ka est étudiée. L'emploi de la bande Ka introduit cependant une grande variabilité de la qualité de réception au niveau des utilisateurs finals, rendant nécessaire l'utilisation d'un protocole de transport mettant en oeuvre des mécanismes spécifiques. Selon une fonction de coût définie, la comparaison des solutions basées sur IP Multicast classiquement utilisées montre que l'utilisation d'une approche hybride couplant l'utilisation des réseaux satellites et terrestres est avantageuse. Le principe de la proposition, nommée Hybrid Satellite Terrestrial Reliable Multicast, consiste ainsi à choisir, en fonction de la taille du groupe, le moyen de diffusion le plus rentable - au vu d'une fonction de coût définie. Une description détaillée de la proposition inclut le comportement de la source et des récepteurs, et le format des messages échangés. Bien que le principe de cette approche soit simple, plusieurs points durs sont liés à la conception des mécanismes adéquats. Ces problèmes concernent notamment la gestion de la fiabilité (utilisation de code correcteur d'erreur ou FEC), l'estimation de taille de très grands groupes, et la reprise des erreurs par voie terrestre (utilisation de réseaux de pair-à-pairs). Ces mécanismes sont étudiés de manière unitaire afin de déterminer des configurations satisfaisantes, et pour détecter des problèmes de performances. Ces mécanismes étant définis, la proposition de transport a été globalement modélisée, de manière à obtenir une vérification fonctionnelle du service proposé. Le protocole a été décrit au moyen du profil UML temps réel TURTLE. Les résultats de validation ont été obtenus grâce à la chaîne d'outils TTool-RTL, et à CADP. ABSTRACT : This thesis studies issues related to the proposition of large scale reliable multipoint communication services. In this context, the possibility to use a geostationary satellite, emitting in the Ka band, to deploy such a service is analysed. However, the use of the Ka band introduces a high variability of quality of reception. Thus, the use of a transport protocol, implementing specific mechanisms, is mandatory. According to a cost function, the comparison of classical solutions, based on IP Multicast, show that a hybrid approach which uses the terrestrial and the satellite networks is advantageous. Consequently, a protocol named Hybrid Satellite Terrestrial Reliable Multicast is proposed. Its principle consists of choosing, depending on the group size, the more profitable network (i.e. terrestrial or satellite network) to transmit information. This choice is made according to a predefined cost function. A sharp description of the proposition, including the hosts' behaviours and the message set-up, is depicted. In spite of the simplicity of the approach, several obstacles appear when one tries to design appropriate mechanisms. These issues include reliability (use of forward error correction), large group size estimation, and terrestrial error recovery (use of peer-topeer networks). Those mechanisms are studied separately to determine satisfactory configurations, and to detect performance issues. After the definition of those mechanisms, the proposition is globally modelized in order to start the formal validation of the proposed service. The model is realized using the real-time UML profile TURTLE, and the validation results are obtained thanks to the TTool-RTL toolkit, and to Aldebaran

A novel multimedia adaptation architecture and congestion control mechanism designed for real-time interactive applications

PhDThe increasing use of interactive multimedia applications over the Internet has created a problem of congestion. This is because a majority of these applications do not respond to congestion indicators. This leads to resource starvation for responsive flows, and ultimately excessive delay and losses for all flows therefore loss of quality. This results in unfair sharing of network resources and increasing the risk of network ‘congestion collapse’. Current Congestion Control Mechanisms such as ‘TCP-Friendly Rate Control’ (TFRC) have been able to achieve ‘fair-share’ of network resource when competing with responsive flows such as TCP, but TFRC’s method of congestion response (i.e. to reduce Packet Rate) is not ideally matched for interactive multimedia applications which maintain a fixed Frame Rate. This mismatch of the two rates (Packet Rate and Frame Rate) leads to buffering of frames at the Sender Buffer resulting in delay and loss, and an unacceptable reduction of quality or complete loss of service for the end-user. To address this issue, this thesis proposes a novel Congestion Control Mechanism which is referred to as ‘TCP-friendly rate control – Fine Grain Scalable’ (TFGS) for interactive multimedia applications. This new approach allows multimedia frames (data) to be sent as soon as they are generated, so that the multimedia frames can reach the destination as quickly as possible, in order to provide an isochronous interactive service. This is done by maintaining the Packet Rate of the Congestion Control Mechanism (CCM) at a level equivalent to the Frame Rate of the Multimedia Encoder.The response to congestion is to truncate the Packet Size, hence reducing the overall bitrate of the multimedia stream. This functionality of the Congestion Control Mechanism is referred to as Packet Size Truncation (PST), and takes advantage of adaptive multimedia encoding, such as Fine Grain Scalable (FGS), where the multimedia frame is encoded in order of significance, Most to Least Significant Bits. The Multimedia Adaptation Manager (MAM) truncates the multimedia frame to the size indicated by the Packet Size Truncation function of the CCM, accurately mapping user demand to available network resource. Additionally Fine Grain Scalable encoding can offer scalability at byte level granularity, providing a true match to available network resources. This approach has the benefits of achieving a ‘fair-share’ of network resource when competing with responsive flows (as similar to TFRC CCM), but it also provides an isochronous service which is of crucial benefit to real-time interactive services. Furthermore, results illustrate that an increased number of interactive multimedia flows (such as voice) can be carried over congested networks whilst maintaining a quality level equivalent to that of a standard landline telephone. This is because the loss and delay arising from the buffering of frames at the Sender Buffer is completely removed. Packets sent maintain a fixed inter-packet-gap-spacing (IPGS). This results in a majority of packets arriving at the receiving end at tight time intervals. Hence, this avoids the need of using large Playout (de-jitter) Buffer sizes and adaptive Playout Buffer configurations. As a result this reduces delay, improves interactivity and Quality of Experience (QoE) of the multimedia application

2003-2004 Louisiana Tech University Catalog

The Louisiana Tech University Catalog includes announcements and course descriptions for courses offered at Louisiana Tech University for the academic year of 2003-2004.https://digitalcommons.latech.edu/university-catalogs/1011/thumbnail.jp