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

Service oriented networking for multimedia applications in broadband wireless networks

Extensive efforts have been focused on deploying broadband wireless networks. Providing mobile users with high speed network connectivity will let them run various multimedia applications on their wireless devices. In order to successfully deploy and operate broadband wireless networks, it is crucial to design efficient methods for supporting various services and applications in broadband wireless networks. Moreover, the existing access-oriented networking solutions are not able to fully address all the issues of supporting various applications with different quality of service requirements. Thus, service-oriented networking has been recently proposed and has gained much attention. This dissertation discusses the challenges and possible solutions for supporting multimedia applications in broadband wireless networks. The service requirements of different multimedia applications such as video streaming and Voice over IP (VoIP) are studied and some novel service-oriented networking solutions for supporting these applications in broadband wireless networks are proposed. The performance of these solutions is examined in WiMAX networks which are the promising technology for broadband wireless access in the near future. WiMAX networks are based on the IEEE 802.16 standards which have defined different Quality of Service (QoS) classes to support a broad range of applications with varying service requirements to mobile and stationary users. The growth of multimedia traffic that requires special quality of service from the network will impose new constraints on network designers who should wisely allocate the limited resources to users based on their required quality of service. An efficient resource management and network design depends upon gaining accurate information about the traffic profile of user applications. In this dissertation, the access level traffic profile of VoIP applications are studied first, and then a realistic distribution model for VoIP traffic is proposed. Based on this model, an algorithm to allocate resources for VoIP applications in WiMAX networks is investigated. Later, the challenges and possible solutions for transmitting MPEG video streams in wireless networks are discussed. The MPEG traffic model adopted by the WiMAX Forum is introduced and different application-oriented solutions for enhancing the performance of wireless networks with respect to MPEG video streaming applications are explained. An analytical framework to verify the performance of the proposed solutions is discoursed, and it is shown that the proposed solutions will improve the efficiency of VoIP applications and the quality of streaming applications over wireless networks. Finally, conclusions are drawn and future works are discussed

Multimedia streaming over wireless channels

The improvements in mobile communication systems have accelerated the development of new multimedia streaming techniques to increase the quality of streaming data over time varying wireless channels. In order to increase multimedia quality, error control schemes are indispensable due to time-varying and erroneous nature of the channel. However, relatively low channel capacity of wireless channels, and dependency structure in multimedia limit the eectiveness of existing error control schemes and require more sophisticated techniques to provide quality improvement on the streaming data. In this thesis, we propose sender driven multimedia streaming algorithms that incorporate error control schemes of FEC, ARQ, and packet scheduling by considering media and channel parameters such as packet importance, packet dependencies, decoding deadlines, channel state information, and channel capacity. Initially, we have proposed a multi-rate distortion optimization framework so as to jointly optimize FEC rate and packet selection by minimizing end-to-end distortion to satisfy a specified Quality of Service under channel capacity constraint. Minimization of end-to-end distortion causes computational complexity in the rate distortion optimization framework due to dependency in encoded multimedia. Therefore, we propose multimedia streaming algorithms that select packet and FEC rate with reduced computational complexity and high quality as compared with multi-rate distortion optimization framework. Additionally, protocol stack of a UMTS cellular network system with W-CDMA air interface is presented in order to clarify the relation between proposed multimedia streaming algorithms and UMTS system that is used in simulations. Finally, proposed algorithms are simulated and results demonstrate that proposed algorithms improve multimedia quality significantly as compared to existing methods

Robust P2P Live Streaming

Projecte fet en col.laboració amb la Fundació i2CATThe provisioning of robust real-time communication services (voice, video, etc.) or media contents through the Internet in a distributed manner is an important challenge, which will strongly influence in current and future Internet evolution. Aware of this, we are developing a project named Trilogy leaded by the i2CAT Foundation, which has as main pillar the study, development and evaluation of Peer-to-Peer (P2P) Live streaming architectures for the distribution of high-quality media contents. In this context, this work concretely covers media coding aspects and proposes the use of Multiple Description Coding (MDC) as a flexible solution for providing robust and scalable live streaming over P2P networks. This work describes current state of the art in media coding techniques and P2P streaming architectures, presents the implemented prototype as well as its simulation and validation results

Source video rate allocation and scheduling policy design for wireless networks

In this work we address the problem of designing an efficient algorithm that allows high quality real time video streaming, from the source coding and scheduling perspectives. We take into account some key quantities like frame correlation, channel constraints and QoS metrics such as distortion and PSNR. We first formalize the problem in a mathematical fashion and then propose a Markov-chain based solution that applies to quantized values of the metrics involved in the framewor

An experimental study of VBR video over various ATM switch architectures

One of the most important components of an Asynchronous Transfer Mode (ATM) network is the switch. Switch design is not a part of the ATM standards so vendors use a wide variety of techniques to build their switches. In this paper, the authors present experimental results of switching and multiplexing real-time Variable Bit Rate (VBR) video traffic (JPEG, MPEG-1, and MPEG-2) through two different ATM switch architectures. Real-time VBR traffic, such as digital video, is particularly interesting due to its high demands in terms of bandwidth, real-time delivery and processing requirements. The experiments show that the fastest switches, i.e., lowest latencies, do not necessarily perform better when transmitting VBR video. The impact of the high speed network components; characteristics, such as switch fabric architecture, buffering strategies, and higher layer transport protocols (i.e., UDP, TCP/IP), are illustrated through the experimental results

Single queue priority scheduler for video transmission in IEEE 802.11 networks

Includes bibliography.Mobile video transmission poses many challenges in standard wireless network like Wireless Local Area Network or IEEE 802.11. The challenges range from handover, delay, packet loss, jitter, fading and signal loss. Some studies have suggested an increase in network resources as a way to cater for the huge demands and reduce congestion in the network, while others suggest that optimizing the available resources might also reduce these challenges. In line with the optimization approach, this study proffers a solution to the video loss in IEEE 802.11 networks. It uses the Single-Queue Priority Scheduler to rearrange the video frames based on their importance. An MPEG frame (trace file) was rearranged by assigning weights to the video frames I, B and P. These frames were then prioritized and arranged in a single queue. A parameter to actively arrange the queue ( ) was deduced from three metrics- deadline, priority and cost. This value Sn was used to arrange the video trace or frames from the lowest to the highest. The arranged video trace or frames were injected into the queue and transmitted in that order. The results show that the implementation of Single-Queue Priority Scheduler algorithm improves the video transmission in Wireless Local Area Network. Without Single-Queue Priority Scheduler algorithm, the buffer overflow loss is 22.8% of the total load, but with SQPS algorithm, it is 8% of the total load. Without SQPS algorithm, the Packet Loss Ratio is about61%; but with the SQPS algorithm, the PLR reduces to 34%. Although, this scheduling algorithm produced better results with a reduction in packet loss, there were still some losses in the network