vSkyConf: Cloud-assisted Multi-party Mobile Video Conferencing

As an important application in the busy world today, mobile video conferencing facilitates virtual face-to-face communication with friends, families and colleagues, via their mobile devices on the move. However, how to provision high-quality, multi-party video conferencing experiences over mobile devices is still an open challenge. The fundamental reason behind is the lack of computation and communication capacities on the mobile devices, to scale to large conferencing sessions. In this paper, we present vSkyConf, a cloud-assisted mobile video conferencing system to fundamentally improve the quality and scale of multi-party mobile video conferencing. By novelly employing a surrogate virtual machine in the cloud for each mobile user, we allow fully scalable communication among the conference participants via their surrogates, rather than directly. The surrogates exchange conferencing streams among each other, transcode the streams to the most appropriate bit rates, and buffer the streams for the most efficient delivery to the mobile recipients. A fully decentralized, optimal algorithm is designed to decide the best paths of streams and the most suitable surrogates for video transcoding along the paths, such that the limited bandwidth is fully utilized to deliver streams of the highest possible quality to the mobile recipients. We also carefully tailor a buffering mechanism on each surrogate to cooperate with optimal stream distribution. We have implemented vSkyConf based on Amazon EC2 and verified the excellent performance of our design, as compared to the widely adopted unicast solutions.Comment: 10 page

ChitChat: Making Video Chat Robust to Packet Loss

Video chat is increasingly popular among Internet users. Often, however, chatting sessions suffer from packet loss, which causes video outage and poor quality. Existing solutions however are unsatisfying. Retransmissions increase the delay and hence can interact negatively with the strict timing requirements of interactive video. FEC codes introduce extra overhead and hence reduce the bandwidth available for video data even in the absence of packet loss. This paper presents ChitChat, a new approach for reliable video chat that neither delays frames nor introduces bandwidth overhead. The key idea is to ensure that the information in each packet describes the whole frame. As a result, even when some packets are lost, the receiver can still use the received packets to decode a smooth version of the original frame. This reduces frame loss and the resulting video freezes and improves the perceived video quality. We have implemented ChitChat and evaluated it over multiple Internet paths. In comparison to Windows Live Messenger 2009, our method reduces the occurrences of video outage events by more than an order of magnitude

Congestion control for real-time interactive multimedia streams

The Internet is getting richer, and so the services. The richer the services, the more the users demand. The more they demand, the more we guarantee(1). This thesis investigates the congestion control mechanisms for interactive multimedia streaming applications. We start by raising a question as to why the congestion control schemes are not widely deployed in real-world applications, and study what options are available at present. We then discuss and show some of the good reasonings that might have made the control mechanism, specifically speaking the rate-based congestion control mechanism, not so attractive. In an effort to address the problems, we identify the existing problems from which the rate-based congestion control protocol cannot easily escape. We therefore propose a simple but novel windowbased congestion control protocol that can retain smooth throughput property while being fair when competing with TCP, yet still being responsive to the network changes. Through the extensive ns-2 simulations and the real-world experiments, we evaluate TFWC, our proposed mechanisms, and TFRC, the proposed IETF standard, in terms of network-oriented metrics (fairness, smoothness, stability, and responsive), and end-user oriented metrics (PSNR and MOS) to throughly study the protocol’s behaviors. We then discuss and conclude the options of the evaluated protocols for the real application. (1)We as congestion control mechanisms in the Internet

A QoE-driven Vertical Handover Management Framework for Multimedia Services over Wireless Networks

With advances in wireless technology and mobile devices, the number of mobile users using multimedia services has increased significantly in recent years. Mobile devices can be connected and roam on heterogeneous wireless networks. The IEEE 802.21 group has designed a Media Independent Handover (MIH) standard to ensure seamless Vertical Handover (VHO) in heterogeneous networks. However, the standard currently depends on features of the network (e.g. the type of network and available bandwidth) to achieve seamless VHO. This approach is limited, as it does not consider how a Quality of Experience (QoE) can be provided and maintained for customers when delivering multimedia services in heterogeneous wireless networks. The aim of the project is to develop a novel QoE-driven VHO management framework for providing and maintaining an appropriate level of QoE of multimedia services as the mobile user’s actual requirements in heterogeneous wireless networks. A QoE-driven VHO algorithm is more efficient for maintaining this acceptable QoE of multimedia services than traditional network-based or QoS-based VHO algorithms. There are three main contributions during this project. Firstly, A thorough evaluation of the performance of voice and video services via Skype was carried out in terms of the QoE metric (i.e. MOS). This work identified the impact of video content and packet loss on the QoE metric for voice and video communication services over wireless networks. Secondly, a QoE-driven VHO algorithm was developed to provide and maintain an acceptable QoE of mobile video services for mobile users. Compared to a traditional network-based VHO algorithm, this algorithm can provide better QoE and maintain acceptable QoE. Lastly, the User-centric QoE-driven (UCQoE) VHO framework to provide satisfactory QoE of multimedia services according to the mobile user’s requirements. The framework allows users to set their own preferences (e.g. quality-guarantee or cost-free) and carry out VHO operations accordingly. The evaluation showed that the proposed framework can provide a better QoE for delivered video services than QoS-based and network-based VHO algorithms. Furthermore, the proposed framework can be used to avoid unnecessary cost of mobile data when the option of cost-free is preferred by the user. During this project, three international conference papers had been published and a journal paper has been submitted to IEEE Transactions on Mobile Computing. The main contribution-UCQoE VHO management framework can be developed to maintain QoE of all mobile services in the future

Skype video responsiveness to bandwidth variations

The TCP/IP stack has been extremely successful for reliable delivery of best-effort, time insensitive elastic type data traffic. Nowadays, the Internet is rapidly evolving to become an equally efficient platform for multimedia content delivery. Key examples of this evolution are, to name few, YouTube, Skype Audio/Video, IPTV, P2P video distribution such as Coolstreaming or Joost. While YouTube streams videos using the Transmission Control Protocol (TCP), applications that are time-sensitive such as Skype VoIP or Video Conferencing employ the UDP because they can tolerate small loss percentages but not delays due to TCP recovery of losses via retransmissions. Since the UDP does not implement congestion control, these applications must implement those functionalities at the application layer in order to avoid congestion and preserve network stability. In this paper we investigate Skype Video in order to discover to what extent this application is able to throttle its sending rate to match the unpredictable Internet bandwidth while preserving resource for co-existing best-effort TCP traffic

Internet traffic classification for high-performance and off-the-shelf systems

Tesis doctoral inédita, leída en la Universidad Autónoma de Madrid, Escuela Politécnica Superior, Departamento de Tecnología Electrónica y de las Comunicaciones, 2013