Live Prefetching for Mobile Computation Offloading

The conventional designs of mobile computation offloading fetch user-specific data to the cloud prior to computing, called offline prefetching. However, this approach can potentially result in excessive fetching of large volumes of data and cause heavy loads on radio-access networks. To solve this problem, the novel technique of live prefetching is proposed in this paper that seamlessly integrates the task-level computation prediction and prefetching within the cloud-computing process of a large program with numerous tasks. The technique avoids excessive fetching but retains the feature of leveraging prediction to reduce the program runtime and mobile transmission energy. By modeling the tasks in an offloaded program as a stochastic sequence, stochastic optimization is applied to design fetching policies to minimize mobile energy consumption under a deadline constraint. The policies enable real-time control of the prefetched-data sizes of candidates for future tasks. For slow fading, the optimal policy is derived and shown to have a threshold-based structure, selecting candidate tasks for prefetching and controlling their prefetched data based on their likelihoods. The result is extended to design close-to-optimal prefetching policies to fast fading channels. Compared with fetching without prediction, live prefetching is shown theoretically to always achieve reduction on mobile energy consumption.Comment: To appear in IEEE Trans. on Wireless Communicatio

Network coding meets multimedia: a review

While every network node only relays messages in a traditional communication system, the recent network coding (NC) paradigm proposes to implement simple in-network processing with packet combinations in the nodes. NC extends the concept of "encoding" a message beyond source coding (for compression) and channel coding (for protection against errors and losses). It has been shown to increase network throughput compared to traditional networks implementation, to reduce delay and to provide robustness to transmission errors and network dynamics. These features are so appealing for multimedia applications that they have spurred a large research effort towards the development of multimedia-specific NC techniques. This paper reviews the recent work in NC for multimedia applications and focuses on the techniques that fill the gap between NC theory and practical applications. It outlines the benefits of NC and presents the open challenges in this area. The paper initially focuses on multimedia-specific aspects of network coding, in particular delay, in-network error control, and mediaspecific error control. These aspects permit to handle varying network conditions as well as client heterogeneity, which are critical to the design and deployment of multimedia systems. After introducing these general concepts, the paper reviews in detail two applications that lend themselves naturally to NC via the cooperation and broadcast models, namely peer-to-peer multimedia streaming and wireless networkin

Adaptive Live Video Streaming by Priority Drop

In this paper we explore the use of Priority-progress streaming (PPS) for video surveillance applications. PPS is an adaptive streaming technique for the delivery of continuous media over variable bit-rate channels. It is based on the simple idea of reordering media components within a time window into priority order before transmission. The main concern when using PPS for live video streaming is the time delay introduced by reordering. In this paper we describe how PPS can be extended to support live streaming and show that the delay inherent in the approach can be tuned to satisfy a wide range of latency constraints while supporting fine-grain adaptation

Software Defined Resource Allocation for Attaining QoS and QoE Guarantees at the Wireless Edge

Wireless Internet access has brought legions of heterogeneous applications all sharing the same resources. However, current wireless edge networks that provide Quality of Service (QoS) guar-antees that only cater to worst or average case performance lack the agility to best serve these diverse sessions. Simultaneously, software reconfigurable infrastructure has become increasingly mainstream to the point that dynamic per packet and per flow decisions are possible at multiple layers of the communications stack. In this dissertation, we explore several problems in the space of cross-layer optimization of reconfigurable infrastructure with the objective of maximizing user-perceived Quality of Experience (QoE) under the resource constraints of the Wireless Edge. We first model the adaptive reconfiguration of system infrastructure as a Markov Decision Pro-cess with a goal of satisfying application requirements, and whose transition kernel is discovered using a reinforcement learning approach. Our context is that of reconfigurable (priority) queueing, and we use the popular application of video streaming as our use case. Self declaration of states by all participating applications is necessary for the success of the approach. This need motivates us to design an open market-based system which promotes the truthful declaration of value (state). We show in an experimental setup that the benefits of such an approach are similar to those of the learning approach. Implementations of these techniques are conducted on off-the-shelf hardware, which have inherent restrictions on reconfigurability across different layers of the network stack. Consequently, we exploit a custom hardware platform to achieve finer grained reconfiguration capabilities like per packet scheduling and develop a platform for implementation and testing of scheduling protocols with ultra-low latency requirements. Finally, we study a distributed approach for satisfying strict application requirements by leveraging end user devices interested in a shared objective. Such a system enables us to attain the necessary performance goals with minimal use of centralized infrastructure

Voice Over Sensor Networks

Wireless sensor networks have traditionally focused on low duty-cycle applications where sensor data are reported periodically in the order of seconds or even longer. This is due to typically slow changes in physical variables, the need to keep node costs low and the goal of extending battery lifetime. However, there is a growing need to support real-time streaming of audio and/or low-rate video even in wireless sensor networks for use in emergency situations and shortterm intruder detection. In this paper, we describe a real-time voice stream-capability in wireless sensor networks and summarize our deployment experiences of voice streaming across a large sensor network of FireFly nodes in an operational coal mine. FireFly is composed of several integrated layers including specialized low-cost hardware, a sensor network operating system, a real-time link layer and network scheduling. We are able to provide efficient support for applications with timing constraints by tightly coupling the network and task scheduling with hardware-based global time synchronization. We use this platform to support 2-way audio streaming concurrently with sensing tasks. For interactive voice, we investigate TDMA-based slot scheduling with balanced bi-directional latency while meeting audio timeliness requirements. Finally, we describe our experimental deployment of 42 nodes in a coal mine, and present measurements of the end-to-end throughput, jitter, packet loss and voice quality