The design and implementation of a multimedia storage server tosupport video-on-demand applications

In this paper we present the design and implementation of a client/server based multimedia architecture for supporting video-on-demand applications. We describe in detail the software architecture of the implementation along with the adopted buffering mechanism. The proposed multithreaded architecture obtains, on one hand, a high degree of parallelism at the server side, allowing both the disk controller and the network card controller work in parallel. On the other hand; at the client side, it achieves the synchronized playback of the video stream at its precise rate, decoupling this process from the reception of data through the network. Additionally, we have derived, under an engineering perspective, some services that a real-time operating system should offer to satisfy the requirements found in video-on-demand applications.This research has been supported by the Regional Research Plan of the Autonomus Community of Madrid under an F.P.I. research grant.Publicad

Writing an Efficient Device Driver for a Multimedia Teleconferencing System

Modern high speed networks, such as ATM, can provide the bandwidth and the QoS guarantees to demanding real-time multimedia applications. However, overall performance of a networked multimedia application will greatly depend on the in-host data movement. Analyzing the characteristics and requirements of those applications, we came to several conclusions about the operation of the multimedia devices' drivers. We applied these conclusions in the design and implementation of a device driver for a multimedia teleconferencing system, based on IBM RS/6000 servers, running the AIX 3.2 operating system. Tracing the complete in-host data path, we found that though our device driver minimized the movement of data between the teleconferencing card and user main memory, the UDP/IP stack proved to be a cause of delay in the movement of data between user main memory and the network interface. (Also cross-referenced as UMIACS-TR-96-18

BIBS: A Lecture Webcasting System

The Berkeley Internet Broadcasting System (BIBS) is a lecture webcasting system developed and operated by the Berkeley Multimedia Research Center. The system offers live remote viewing and on-demand replay of course lectures using streaming audio and video over the Internet. During the Fall 2000 semester 14 classes were webcast, including several large lower division classes, with a total enrollment of over 4,000 students. Lectures were played over 15,000 times per month during the semester. The primary use of the webcasts is to study for examinations. Students report they watch BIBS lectures because they did not understand material presented in lecture, because they wanted to review what the instructor said about selected topics, because they missed a lecture, and/or because they had difficulty understanding the speaker (e.g., non-native English speakers). Analysis of various survey data suggests that more than 50% of the students enrolled in some large classes view lectures and that as many as 75% of the lectures are played by members of the Berkeley community. Faculty attitudes vary about the virtues of lecture webcasting. Some question the use of this technology while others believe it is a valuable aid to education. Further study is required to accurately assess the pedagogical impact that lecture webcasts have on student learning

Quality assessment and usage behavior of a mobile voice-over-IP service

Voice-over-IP (VoIP) services offer users a cheap alternative to the traditional mobile operators to make voice calls. Due to the increased capabilities and connectivity of mobile devices, these VoIP services are becoming increasingly popular on the mobile platform. Understanding the user's usage behavior and quality assessment of the VoIP service plays a key role in optimizing the Quality of Experience (QoE) and making the service to succeed or to fail. By analyzing the usage and quality assessments of a commercial VoIP service, this paper identifies device characteristics, context parameters, and user aspects that influence the usage behavior and experience during VoIP calls. Whereas multimedia services are traditionally evaluated by monitoring usage and quality for a limited number of test subjects and during a limited evaluation period, this study analyzes the service usage and quality assessments of more than thousand users over a period of 120 days. This allows to analyze evolutions in the usage behavior and perceived quality over time, which has not been done up to now for a widely-used, mobile, multimedia service. The results show a significant evolution over time of the number of calls, the call duration, and the quality assessment. The time of the call, the used network, and handovers during the call showed to have a significant influence on the users' quality assessments

Deployment of Digital Video and Audio Over Electrical SCADA Networks

With the arrival of new hardware and software technologies, supervisory control and data acquisition human-machine interfaces (SCADA/HMI), usually text-based, can now benefit from the advantages the inclusion of multimedia information brings. However, due to the special requirements imposed by such systems, integrating audio and video data into the SCADA interfaces is not a trivial task. In this document we analyze those special characteristics and propose solutions so this integration is possible in power systems communication.Ministerio de Ciencia y Tecnología TIC2000-0367-P4-0

Mobile Communication Networks and Digital Television Broadcasting Systems in the Same Frequency Bands – Advanced Co-Existence Scenarios

The increasing demand for wireless multimedia services provided by modern communication systems with stable services is a key feature of advanced markets. On the other hand, these systems can many times operate in a neighboring or in the same frequency bands. Therefore, numerous unwanted co-existence scenarios can occur. The aim of this paper is to summarize our results which were achieved during exploration and measurement of the co-existences between still used and upcoming mobile networks (from GSM to LTE) and digital terrestrial television broadcasting (DVB) systems. For all of these measurements and their evaluation universal measurement testbed has been proposed and used. Results presented in this paper are a significant part of our activities in work package WP5 in the ENIAC JU project “Agile RF Transceivers and Front-Ends for Future Smart Multi-Standard Communications Applications (ARTEMOS)”