In this paper, we present a watchdog blockchain network (WDN) that allows for simultaneous scalability and decentralization enhancements. Because of the blockchain trilemma, achieving high levels of security, scalability, and decentralization simultaneously is challenging. Typically, public blockchains are fully decentralized at the expense of scalability, whereas private blockchains achieve high scalability at the expense of decentralization. A WDN is proposed to compensate for the low-level decentralization of a private blockchain while maintaining its high-level scalability. A WDN may prevent arbitrary forgery, alteration, and deletion of distributed ledgers in private blockchains owing to weak decentralization. However, the transaction throughput gap between the WDN and the private blockchain can be a significant obstacle in operating a WDN with a private blockchain. Therefore, the WDN in this study has been designed to back up data on the private blockchain in real time over a distributed network environment. The WDN features a network-aware consensus protocol that can quickly complete the consensus process of the three phases. To this end, modern network technologies such as GossipSub, quick UDP internet connections (QUIC), and Google remote procedure call (GRPC) have been appropriately adopted for the WDN. To demonstrate the system performance of the WDN, a blockchain system was fully implemented in the form of docker containers. Furthermore, we verified the feasibility of the WDN by developing decentralized applications (DApps).

In this paper, we propose a resilient and fast block transmission system for Hyperledger Fabric in multi-cloud environments. The goal of the proposed system is to improve the scalability, transaction throughput, and resilience of Hyperledger Fabric by minimizing the block synchronization time among nodes. To achieve this goal, the proposed system is designed to deliver blocks quickly and reliably to all the participating nodes in time-varying multi-cloud environments. The proposed system includes the delay estimating process with O(N) control message overhead over the P2P network, the effective bandwidth estimating process for block transmission, the Gaussian Mixture Model-based clustering and cluster leader selecting process, and hybrid P2P multicast tree constructing process. In addition, a control message format and delivery process are proposed to efficiently provide hybrid P2P multicast tree and neighbor nodes information to all the participating nodes. And we propose a pull-based local block loss recovery process that can receive lost blocks from multi-node without complicated scheduling using a rateless code. The proposed system is fully implemented by using well-known open sources (e.g., Hyperledger Fabric, Docker, Containernet, and Mininet) and Go/C/Python. Experiment results show that the proposed system can reduce the maximum block arriving time among all the participating nodes by approximately 50%~95% compared to the existing algorithms. This improves not only blockchain transaction per second, but also resilience to various network-layer vulnerabilities and attacks that may occur when the block propagation delay increases.

This paper presents a novel HTTP adaptive augmented reality (AR) streaming system. The goal of the proposed system is to provide high quality AR streaming services with low latency through a time-varying wireless network. In the proposed system, we employ a progressive mesh technique to enhance the scalability of AR streaming services, and adopt the metafile structure to support various attributes of progressive meshes necessary for rendering, displaying, and scheduling according to wireless network conditions. Based on this metafile, the proposed system requests the next chunk among AR contents to improve human visual perceptual quality considering the displaying scale values of AR contents on the screen. The proposed system is fully implemented by using well-known open sources (e.g., Android, OpenCV, OpenGL, and Nexus) and C/C++. Finally, the system is examined in real wireless network environments.

This paper presents a collaborative virtual 3D object modeling system for mobile augmented reality (AR) streaming services over 5G networks. The objective of the proposed system is to provide AR streaming services with low latency and high virtual object quality by effectively leveraging mobile edge cloud (MEC) and device-to-device (D2D) communication. MEC is utilized to perform computation-intensive 3D object modeling, such as virtual object slicing and mesh simplification, with abundant computing and storage resources. D2D communication is also utilized to reduce transmission delay by sharing virtual objects of interest among adjacent users. To achieve this goal, we propose a part-segment quality selection algorithm that can control the quality of each part segment and the transmission source by considering the user's network condition. In the proposed system, all elements in the AR cloud, MEC, and mobile device are technically integrated to work together seamlessly. Finally, the proposed system is fully implemented using well-known open-source frameworks, including WebGL and AR.js. The system was then examined in real wireless network environments. The experimental results demonstrate the performance of the proposed system. The proposed system outperforms conventional systems in terms of service latency and visual 3D object quality.

Web 3.0 is an emerging Internet paradigm based predominantly on the blockchain technology. Because Web 3.0 applications are designed to operate over trustless and permissionless networks, they can have significant advantages, such as decentralized control structures and transparency. Hence, existing web applications are being reproduced using Web 3.0 technologies. In contrast, real-time services are still implemented with the Web 2.0 architecture. In particular, implementing Web 3.0 media streaming requires modifications to the service architecture of existing media streaming systems because some technical difficulties exist. For example, as data moves from centralized data centers to distributed storage, the user's quality of experience may be severely degraded. In addition, the software components comprising the Web 3.0 stack, such as interplanetary file system, cryptocurrency wallets, and the Ethereum JavaScript API are not compatible with various combinations of OSs, media players, and browsers. Therefore, in this study, we propose an end-to-end system architecture designed for Web 3.0 real-time services, which prevents degradation of service quality. Further, we present a media NFT marketplace named Retriever (https://retriever.live) fully developed using Web 3.0 technologies. Retriever allows users to enjoy watching video content and further to directly trade their content without intermediaries by ensuring the privacy of the data and managing digital intellectual property. In particular, Retriever does not sacrifice the user experience and is compatible with multiple mobile devices.

This work presents a novel HTTP adaptive streaming system to maximize overall video streaming service quality of all the clients over SDN-enabled Wi-Fi APs. To archive the goal, the proposed system employs multipath technology to overcome the limitation of a Wi-Fi AP, and SDN technology to effectively harmonize multiple Wi-Fi APs. During the streaming services, SDN controller periodically monitors the traffic information collected from the APs, and then determines video bitrate and segment transmission schedule based on this information. The proposed system is fully implemented in real Wi-Fi network environments.

This paper presents an effective adaptive 360° video VR streaming system based on the Spatial Representation Description (SRD) feature of MPEG-DASH. The goal of the proposed system is to provide a seamless 360° video VR streaming service with high spatial quality over the time-varying wireless network. To achieve this goal, the proposed system adopts tile-based viewport-adaptive streaming that is a promising way to efficiently deliver 360° video. In accordance with the SRD information and user’s head movement, the proposed system is designed to enhance the quality of the user experience by intensively allocating a limited bit budget to specific segments of interest without noticeable blanks. Additionally, we propose a fast recovery algorithm to minimize the effect of blanks caused by unpredictable head movement. Moreover, the proposed system considers the binocular disparity of the human eye to improve the subjective quality of stereo video. The proposed system is fully implemented by using well-known open sources (e.g., OpenHMD, Kvazaar and GPAC), C/C++, Python and head-mounted display (HMD) equipment. Finally, the experimental results are provided to verify the performance of the proposed system in real wireless network environments.

This paper presents a quick block transport system for scalable Hyperledger Fabric (HLF) blockchain over device-to-device (D2D)-assisted 5G networks. The objective of the proposed system is to increase the transaction throughput and further enable the application of the blockchain to delay-sensitive 5G services. To this end, the system rapidly disseminates the block data to all the peers on the blockchain network. The blockchain network is organized with mobile edge cloud (MEC) server and mobile peers in the 5G network. To ensure a scalable and reliable blockchain architecture, the core functionalities of the HLF blockchain are appropriately distributed among the mobile edge cloud and each mobile peer. The MEC server handles core functionalities for the blockchain network, such as block generation and membership management. In addition, it is responsible for gathering the network status of all peers and organizing the overlay structure to be utilized to rapidly propagate the block with a low network cost. On the other hand, the mobile peers handle the endorsing and validating functionalities to prevent block data tampering in the blockchain network. Furthermore, locality-aware device-to-device networking is utilized in 5G network coverage to achieve faster block dissemination. The proposed system is designed to work seamlessly in an existing HLF blockchain framework with consideration of scalability and compatibility. It is fully implemented using the open-source HLF and OpenAirInterface. To verify the effectiveness of the proposed block transport system, it was evaluated in an actual 5G network as well as 802.11 network. The experimental results verified its feasibility.

This work presents a progressive mesh-based HTTP adaptive augmented reality (AR) streaming system that provides high quality AR streaming services with low latency over a time-varying wireless network. In the proposed system, a progressive mesh is employed to support the scalability of AR streaming services, a novel meta-file is proposed to efficiently provide the rate and view-dependent quality information. By using this meta-file, the proposed system determines the next request chunk to enhance human visual perceptual quality considering the view-dependent projected images of AR objects on the screen. The proposed system is realized using open sources (e.g., Android, OpenCV, OpenGL, and Nexus). Finally, experimental results are given to verify the performance improvement of the proposed system.

This work presents a cloud-assisted augmented reality (AR) streaming service system over wireless network. The goal of the proposed system is to provide interactive AR streaming services of high quality with low latency through a time-varying wireless network. In the proposed system, we employ progressive mesh technique to enhance the scalability of AR streaming services, and adopt meta-file structure to include various attributes of progressive meshes of AR contents necessary for rendering and displaying according to the wireless network conditions. Based on this meta-file provided by a cloud AR server, the proposed system requests the next chunk among AR contents to improve the human visual perceptual quality of all selected AR contents while considering the marker sizes detected by the camera of the mobile AR device. The proposed system is fully implemented by using well-known open sources (e.g., Android, ARCore, and Nexus) and C/C++. Finally, the system is examined in real wireless network environments.

Summary This paper proposes a novel congestion‐aware HTTP adaptive streaming (HAS) system to improve video quality of clients over software‐defined network (SDN)–enabled Wi‐Fi network. The proposed system employs SDN technology to efficiently control the network resources and video quality over the competing Wi‐Fi network. In the proposed system, an SDN‐enabled access point (AP) can monitor the streaming status of clients, such as the buffered playback time and information of requested segments. An SDN controller collects not only the streaming status of clients but also the Wi‐Fi network conditions from the AP using OpenFlow. Additionally, when a congestion is going to occur, it determines the segment bitrate of each client to provide seamless and high‐quality streaming services. The proposed system is fully implemented using open‐source programs, such as ONOS and hostapd, and is examined in a real Wi‐Fi network. The experimental results verify that the proposed system can provide seamless and high‐quality HAS services for the clients compared with existing systems.

This work presents a cooperative and distributive caching system over the information-centric networking (ICN) tree. The goal of the proposed caching system is to provide seamless video streaming services with high network utilization over the ICN tree. The proposed system employs a metafile made by a scalable caching scheme for simple but effective Content Store (CS) management, and the caching information among ICN nodes is delivered by modified ICN interest/data packet headers with no additional control overhead. Based on the delivered caching information and the metafile, each ICN node determines its own caching chunks in a distributive way to minimize the required peak bandwidth for seamless streaming. The proposed caching system is fully implemented using the NDNSim for large scale simulation. Moreover, a small-scale real cellular wireless network testbed is realized by using CCNx and Raspberry PIs.

In this work, we propose an energy-efficient HTTP adaptive streaming with hybrid TCP/UDP over heterogeneous wireless networks. Unlike the existing HTTP adaptive streaming system, the proposed system can utilize both TCP and UDP to overcome the performance degradation problem of TCP and improve the energy efficiency. The proposed system handles the types of transport protocols, requested video quality, and amount of data received through each wireless network as control variables based on network conditions, buffer occupancy, and energy condition of mobile terminal. When UDP is selected, the proposed system employs Raptor code as a forward error correction scheme to support a reliable data delivery, and adaptively chooses the proper Raptor code parameters such as symbol size and amount of redundant data, and requests redundant data from the server to reduce the overhead. The proposed system is fully implemented in an Android-based mobile terminal and tested in real WiFi and LTE networks. The experiment results show that the proposed system can achieve good energy efficiency while providing a seamless high-quality HTTP adaptive streaming service.

This paper presents a cooperative base station caching and X2 link traffic offloading system for video streaming services over software-defined networking (SDN)-enabled 5G networks. The goal of the proposed system is to provide seamless video streaming services to users while alleviating the backhaul traffic in the cellular network by effectively caching contents at the base stations and appropriately offloading parts of the backhaul traffic to X2 links. In the proposed system, the main functionalities of content-centric networking (CCN) are deployed at the base stations to take advantage of the attributes offered by CCN such as in-network caching and name-based forwarding. Furthermore, SDN technology is adopted not only to solve the practical issues of CCN, but also to control network resources efficiently. The proposed system is designed to determine near optimal caching positions and routes for all content in two phases to reduce the computational complexity for the real-time processing. The proposed system is fully implemented using well-known open sources, e.g., CCNx and ONOS, and C/C++. Finally, the system is examined in real wireless network environments. During the experiment, the proposed system can reduce cellular traffic by more than 13 percent over existing traffic offloading algorithms.

This work presents a mobile data offloading system for video streaming services over software-defined networking (SDN)-enabled wireless networks. The goal of the proposed system is to alleviate cellular network congestion by offloading parts of video traffic to a WiFi network while improving video quality of all users by efficiently and fairly sharing the limited long term evolution (LTE) resources. In the proposed system, SDN architecture is applied to the wireless network environment to quickly react to time-varying network conditions and finely control the amount of traffic transmitted through LTE and WiFi networks. Under the SDN-enabled wireless environment, we frame the mobile data offloading problem for video streaming services as an asymmetric Nash bargaining game to address conflict among competitive mobile users. Furthermore, we propose a resource allocation algorithm that pursues an effective trade-off between global system utility and quality-of-service fairness among users. The system is fully implemented using ONOS SDN controller and Raspberry PI-3-based mobile devices, and performance is evaluated over real wireless networks.

This work presents a video quality-aware traffic steering system that provides stable video streaming services to users by steering part of the macrocell traffic into small cells. The proposed system achieves a good balance between fairness and social welfare in terms of the video quality by allocating the radio resource of the macro base station. The user data flow is split into two parts toward the macro base station and small cell AP, and the users simultaneously receive their data from both. In the proposed system, fountain code is employed to achieve dual connectivity enhancement by overcoming practical issues in the dual connectivity architecture. Furthermore, SDN technology is adopted not only to quickly react to time-varying network status but also to control network resources efficiently. The proposed resource allocation process is designed to require a low running time complexity. The proposed system is implemented using C/C++, Java, and a well-known ONOS open source. The experimental results verify that the proposed system can achieve much better performance than other traffic steering algorithms.

This paper presents a reliable and energy-efficient hybrid screen mirroring multicast system for sharing high-quality real-time multimedia service with adjacent mobile devices over WiFi network. The proposed system employs overhearing-based multicast transmission scheme with Raptor codes and NACK-based retransmission to overcome well-known WiFi multicast problems such as low transmission rate and high packet loss rate. Furthermore, to save energy on mobile devices, the proposed system not only shapes the screen mirroring traffic, but also determines the target sink device and Raptor encoding parameters such as the number of source symbols, symbol size, and code rate while considering the energy consumption and processing delay of the Raptor encoding and decoding processes. The proposed system is fully implemented in Linux-based single board computers and examined in real WiFi network. Compared to existing systems, the proposed system can achieve good energy efficiency while providing a high-quality screen mirroring service.

This paper presents a game-theoretic scalable offloading system that provides seamless video streaming services by effectively offloading parts of video traffic in all video streaming services to a WiFi network to alleviate cellular network congestion. The system also consolidates multiple physical paths in a cost-effective manner. In the proposed system, the fountain encoding symbols of compressed video data are transmitted through long term evolution (LTE) and WiFi networks concurrently to flexibly control the amount of video traffic through the WiFi network as well as mitigate video quality degradation caused by wireless channel errors. Furthermore, the progressive second price auction mechanism is employed to allocate the limited LTE resources to multiple user equipment in order to maximize social welfare while converging to the ε-Nash equilibrium. Specifically, we design an application-centric resource valuation that explicitly considers both the realistic wireless network conditions and characteristics of video streaming services. In addition, the scalability and convergence properties of the proposed system are verified both theoretically and experimentally. The proposed system is implemented using network simulator 3. Simulation results are provided to demonstrate the performance improvement of the proposed system.

This paper presents an effective mobile cloud-based interactive 3D rendering and streaming system with data-service cost constraints over heterogeneous wireless networks. The proposed system contains software-defined networking (SDN)-enabled adaptive cloud resource management module and context-aware 3D rendering and streaming module to enhance the quality-of-service of interactive 3D rendering and streaming services. The first module is designed to efficiently control the limited cloud resources, such as server computing power and available backbone link bandwidth, and the second module is implemented to pursue an effective tradeoff between 3D rendering quality and compressed image quality. The system is implemented using Mininet network emulator with an OpenDay-light SDN controller, and VirtualGL with an online open-source 3D game, PlaneShift. Finally, the system is examined at a mobile device connected to real long-term evolution and WiFi networks.

This paper presents a smart base station-assisted partial-flow device-to-device offloading system that provides seamless video streaming services to clients by effectively offloading parts of the video traffic to D2D networks in order to alleviate the cellular network traffic load. In the proposed system, the main functionalities of the content centric networking (CCN) technology are deployed for efficient and fast cached content management at the base stations and in the devices. First, a logical overlay Chord-like network is constructed among the smart base stations for effectively handling significant numbers of the distributed contents. Then, content management protocols over the Chord-like network are implemented for rapidly finding the D2D server that contains the cached content requested by the D2D client. A two-stage PID-based LTE traffic controller is proposed to determine the amount of traffic to be offloaded to the D2D network among the cellular operator, the D2D servers, and the D2D clients. The proposed system is fully implemented using a CCNx open source and C/C++. Experimental results are provided to demonstrate the performance improvement of the proposed system.

This work presents a control-theoretic scalable device-to-device offloading system that provides seamless video streaming services to clients by effectively offloading parts of the video traffic to D2D networks in order to alleviate the cellular network traffic load. In the proposed system, the main functionalities of the content centric networking (CCN) technology are employed to simultaneously download the video content from multiple wireless networks. A two-stage PID-based LTE traffic controller is proposed to determine the amount of traffic to be offloaded to the D2D network among the cellular operator, the D2D servers, and the D2D clients. The proposed system is fully implemented using a CCNx open source and C/C++. Experimental results are provided to demonstrate the performance improvement of the proposed system.

In this work, we propose a Raptor codes-based energy-efficient screen mirroring system. To mitigate the video quality degradation caused by wireless network error, the proposed system employs systematic Raptor codes as forward error correction scheme. For energy saving at the mobile device, the proposed system not only shapes screen mirroring traffic, but also determines the Raptor encoding parameters including the symbol size, the number of source symbols, and the code rate. The proposed system is fully implemented on single board computers, and examined over real wireless network.

This work presents a QoE-based interworking call admission, handoff, and packet scheduling system to provide a seamless multimedia streaming services over LTE network. First, QoE metrics are defined based on the amount of buffered frames at the video decoder buffer. Based on these QoE metrics, the proposed call admission algorithm estimates the cell idleness at the next call arriving time to determine the admission of a new UE and the proposed handoff algorithm determines the appropriate handoff trigger timing to keep a balance of the QoE satisfaction levels among immediately adjacent cells. The proposed packet scheduling algorithm has designed in consideration of both the QoE satisfaction levels and the wireless link states of all admitted UEs. Finally, experimental results indicate that the proposed interworking system improves the QoE satisfaction levels of all admitted UEs and overall network utilization better than other existing methods.

This work presents an efficient adaptive tile partitioning algorithm for HEVC decoder parallelization, which divides a video frame into tiles that are processed independently and concurrently on multi-cores. The proposed algorithm is designed to not only increase decoding performance and energy efficiency, but also enhance the video quality by efficiently balancing the workloads of tiles among cores. Experimental results show ~15% improvements of decoding speedup and ~12% energy savings on a 16-core system. Furthermore, it provides better PSNR than other algorithms during experiment.

In this work, we present an avatar mobility and network condition-aware 3D game over wireless networks. The proposed system is designed to pursue an effective tradeoff between 3D rendering quality and compressed image quality. By considering user-preferred networking cost (i.e., data transmission charges), 3D rendering level and image compression parameters are determined dynamically in order to enhance the quality of service for interactive 3D game. The system is implemented using VirtualGL and an online open-source 3D game, PlaneShift, and examined using a mobile device connected to LTE and WiFi networks.

In this work, we present a SDN-based adaptive cloud network management for 3D rendering and streaming services. In the proposed system, SDN is adopted to provide flexible and intelligent control of cloud network resources. By considering server load and network link states simultaneously, 3D graphic rendering servers, encoding servers, and data transmission paths are dynamically determined with low computational complexity. The proposed system is implemented using the OpenDaylight SDN controller and Mininet network emulator. Finally, simulation results are provided to demonstrate the performance of the proposed system.

In this work, we propose an energy-efficient HTTP adaptive streaming for high-quality video over heterogeneous networks (HetNets). To support high-quality video and overcome the limitations of a single network, the proposed system downloads the video segments by concurrently using the HetNets. In the proposed system, the segment bitrate, number of requested segments, network sleep time, and size of requested data through each wireless network are determined adaptively to provide seamless high-quality video streaming in an energy and cost efficient way. The proposed system is fully implemented in an Android-based mobile device and tested in an actual wireless network environments.

In this work, we propose a Raptor codes-based energy-efficient multipath multimedia transport protocol over heterogeneous wireless networks (HetNets) in order to provide seamless high-quality video streaming service in an energy efficient way. In the proposed protocol, systematic Raptor codes are employed to mitigate the video quality degradation. For energy saving, the proposed protocol not only shapes a video stream into bursts, but also determines the Raptor encoding parameters including code rate, symbol size, and the number of source symbols for energy saving. The proposed protocol is fully implemented on an Android-based smartphone, and examined over real HetNets.

This work presents a multipath multimedia transport protocol (MPMTP), which exploits path diversity over heterogeneous wireless networks. The goal of MPMTP is to provide a seamless high-quality video streaming service by using multiple wireless networks simultaneously. In MPMTP, systematic Raptor codes are adopted to mitigate video quality degradation caused by wireless channel errors, as well as to alleviate a head-of-line blocking problem in multipath environments, and their encoding parameters such as code rate, symbol size, and the number of source symbols are determined on the fly by considering the wireless channel state, Raptor encoding and decoding complexity, and receiver video buffer occupancy. Furthermore, MPMTP performs packet scheduling considering not only wireless network conditions but also packet payload characteristics for smooth video playback. The proposed MPMTP is fully implemented in a Linux kernel and examined over real wireless network environments.

This paper presents a multipath multimedia transport protocol (MPMTP), which exploits path diversity over heterogeneous wireless networks. The goal of MPMTP is to provide a seamless high-quality video streaming service by using multiple wireless networks simultaneously. In MPMTP, systematic Raptor codes are adopted to mitigate video quality degradation caused by wireless channel errors as well as to alleviate a head-of-line blocking problem in multipath environments, and their encoding parameters such as code rate, symbol size, and the number of source symbols are determined on the fly by considering the wireless channel state, Raptor encoding and decoding complexity, and receiver buffer occupancy. Furthermore, MPMTP performs packet scheduling considering not only wireless network conditions but also packet payload characteristics for smooth video playback. The proposed MPMTP is fully implemented in a Linux kernel and examined over real wireless network environments.

This paper presents a seamless high-quality HTTP adaptive streaming algorithm that considers wireless network conditions and the energy consumption of a mobile device with networking cost constraints over heterogeneous wireless networks. In the proposed algorithm, the requested video segments are concurrently delivered through multiple wireless networks to overcome the limitations of a single network. The segment quality level, number of requested segments, network inactive interval, and amount of requested data through each wireless network are determined adaptively in order to provide a seamless high-quality video with low energy consumption and networking cost constraints at the mobile device. The proposed algorithm is fully implemented in an Android-based mobile device and tested in an actual wireless network environment.

In this work, we propose a social relation-aware SVC-based video streaming system over Peer-to-Peer network. In the proposed system, peers efficiently control the limited resources because video streaming services require significantly more network resources than other traditional data services. The proposed system is designed to take into account social relations as well as video quality. Social relation information is obtained from Social Network Services, and peer resources are assigned according to social relation-aware friendship and credit models. In addition, SVC is adopted to support various network environments. Finally, experimental results are provided to evaluate the performance of the proposed system.

This work proposes a novel packet urgency metric-based cross-layer design of packet scheduling and routing algorithms to effectively deliver video streaming data over mobile ad hoc networks based on IEEE 802.11 multi-rate service. First, the packet urgency, node urgency, and route urgency are defined on the basis of the residual time to the destination. Furthermore, the one-hop transmission delay of each packet is estimated using a Kalman filter. Next, the packet scheduling algorithm is proposed to minimize the node urgency without unnecessary packet drops based on both these urgency metrics and the estimated transmission delay. The proposed routing algorithm constructs a route to minimize the derivative of route urgency in order to maximize the number of packets delivered within the required end-to-end delay and distribute the entire traffic load. Finally, simulation results are provided to demonstrate the performance of the proposed cross-layer system.

Recently, the mobile cloud computing received much attention since it can provide various and powerful services for the mobile user. The service discovery technology in mobile cloud computing is very important to efficiently provide the mobile services. In this paper, we proposed a novel preference-aware skyline service discovery system in the mobile cloud system. The objective of our system is to discover preferable services in the mobile cloud system. Our proposed mobile cloud system is composed of zone-based overlay network. In the overlay network, querying user generates the query with user preference extracted by a Singular Value Decomposition method from the user's service access history. The generated query is sent to the mobile cloud system, and then the query is handled by the distributed skyline query processing mechanism to discover user preferable services with low networking communication cost in the mobile cloud system. We conduct experiments to evaluate the performance of our proposed system.

In this work, a networking cost effective video streaming system over heterogeneous wireless networks is proposed. The goal of the proposed system is to provide a high-quality video streaming service with minimum networking cost (i.e., data transmission charges). To overcome performance limitations of a single network, multiple physical paths are concurrently exploited, and the number of packets transmitted through selected paths is determined by considering the user preferred video quality and the networking cost for packet transmission. The proposed system is fully implemented on Java and C/C++, and tested over real WiFi and LTE networks. Experimental results show that the proposed system provides a high-quality video streaming service at relatively lower networking cost compared to existing systems.

ABSTRACT In this paper, we present an entire network load‐aware cooperative routing algorithm based on IEEE 802.11 multi‐rate for video streaming over mobile ad hoc networks. The proposed routing algorithm is designed to minimize the consumed time slots while guaranteeing the required time slots at all the pairs of adjacent nodes over the route and the contention neighbors of these nodes to support the route. Furthermore, the proposed routing algorithm can distribute the network loads well over the entire network. This technology is essential because video streaming applications require stringent quality of service and even larger network resources compared with traditional data services, and these demands may dramatically increase the entire network load and/or cause network congestion. Finally, experimental results are provided to show a performance of the proposed routing algorithm. Copyright © 2011 John Wiley & Sons, Ltd.

In this work, we present a QoE-aware mobile IPTV multicast system over WiMAX network. In the proposed system, the adaptive transmission algorithm at a base station determines the control parameters of a fountain-encoded IPTV multicast stream adaptively to the wireless link states of subscribers in order to provide a stable IPTV multicast service with minimum resource usage on WiMAX network, and the channel grouping algorithm at a server makes near-optimal channel grouping based on channel selection preferences in order to pursue an effective tradeoff between the channel zapping time and the processing complexity of a subscriber. Finally, experimental results are provided to show the performance of the proposed system.

Abstract This paper presents a QoS (quality of service) aware routing and power control algorithm consuming low transmission power for multimedia service over mobilead hocnetwork. Generally, multimedia services need stringent QoS over the network. However, it is not easy to guarantee the QoS over mobilead hocnetwork since its network resources are very limited and time‐varying. Furthermore, only a limited amount of power is available at mobile nodes, which makes the problem more challenging. We propose an effective routing and power control algorithm for multimedia services that satisfies end‐to‐end delay constraint with low transmission power consumption. The proposed algorithm supports the required bandwidth by controlling each link channel quality over route in a tolerable range. In addition, a simple but effective route maintenance mechanism is implemented to avoid link failures that may significantly degrade streaming video quality. Finally, performance comparison with existing algorithms is presented in respect to traditional routing performance metrics, and an achievable video quality comparison is provided to demonstrate the superiority of the proposed algorithm for multimedia services over mobilead hocnetwork. Copyright © 2010 John Wiley & Sons, Ltd.

Abstract This paper presents an energy efficient medium access control (MAC) protocol for delay‐sensitive data transmission over wireless sensor network. In general, energy consumption and delay depend on channel monitoring interval (CMI) and data sensing period at each sensor node. Based on this fact, we propose a new preamble structure to effectively advertise CMI and avoid the overhearing problem. In order to pursue an effective tradeoff between energy consumption and delay, we also develop a CMI determining algorithm that searches for a sub‐optimal solution with a low computational complexity in a distributive way. Finally, experimental results are provided to compare the proposed MAC protocol with existing sensor MAC protocols. Copyright © 2010 John Wiley & Sons, Ltd.

This work presents an overlay multicast tree construction algorithm to support stable multimedia service over the Internet. While constructing a multicast tree, it takes into account not only the link delay, but also the peer stability. Since peers actually show dynamic and unstable behavior over peer-to-peer network, it is essential to consider the peer stability. Furthermore, the weighting factor between link delay and peer stability is adaptively controlled. Basically, Genetic algorithm is employed to obtain a near optimal solution with a low computational complexity. Finally, simulation results are provided to show the performance of the proposed algorithm.

In this work, we propose a simple but effective mesh-pull-based P2P video streaming system using Fountain codes with variable symbol sizes for video-on-demand services. The goal of the proposed system is to provide a stable video streaming service of high quality with a low computational complexity and short initial latency. Fountain codes are adopted in the proposed system to simplify the handshaking procedure which causes a large initial latency, and to support a robust video streaming service. The proposed system works by using feedback information to reduce unnecessary encoded symbol generation. In addition, the Fountain code symbol size is continuously adjusted to minimize additional computational overhead required for Fountain encoding/decoding.

In this paper, we propose an effective end-to-end virtual path construction system, which exploits path diversity over heterogeneous wireless networks. The goal of the proposed system is to provide a high quality live video streaming service over heterogeneous wireless networks. First, we propose a packetization-aware fountain code to integrate multiple physical paths efficiently and increase the fountain decoding probability over wireless packet switching networks. Second, we present a simple but effective physical path selection algorithm to maximize the effective video encoding rate while satisfying delay and fountain decoding failure rate constraints. The proposed system is fully implemented in software and examined over real WLAN and HSDPA networks.

In this paper, we propose a Luby transform encoding pattern-aware symbol packetization algorithm to minimize the quality degradation of video streaming service caused by packet losses over wireless network. To achieve this goal, the relationship among Luby transform encoded symbols is analyzed based on Luby transform encoding pattern, and the proposed packetization algorithm is designed to minimize packet loss effects by reducing the dependency among packets conveying Luby transform encoded symbols. Finally, experimental results are provided to show the performance of the proposed algorithm.

This work presents an effective IPTV channel control algorithm that improves the subscribers' QoE for mobile IPTV service over WiMAX network. We consider video quality of TV channels and channel zapping time as the QoE metrics. The proposed algorithm controls the channel distribution state and the target bit rate of TV channels based on the channel preference information to pursue an effective trade-off between channel zapping time and video quality. Finally, experimental results are provided to demonstrate the performance of proposed system.

In this work, we present an effective IPTV system architecture over WiMAX network. Basically, H.264 MGS coding is employed to adaptively transmit IPTV stream over the time-varying wireless environment. The proposed system is designed to provide an enhanced mobile IPTV service to as many subscribers as possible considering their wireless link states. To achieve this goal, the base layer stream is transmitted with the sparsest modulation scheme and the lowest coding rate to provide IPTV service of minimum quality to all subscribers in the cell. And some parts of the enhancement layer stream are transmitted with the modulation scheme and the coding rate selected to pursue an effective tradeoff between the number of subscribers that receive the enhancement layer stream and their IPTV service quality. Finally, experimental results are provided to show the performance of the proposed system.

This work proposes a deployment scheme of fountain code for delay-sensitive video transmission over wireless packet-switching networks. As the number of source symbols increases, the performance of fountain code improves but delay increases. To solve this problem we investigate the design of fountain code and propose an effective source symbol selection algorithm to eliminate predominant correlation to specific packets.

This work presents an effective cross-layer video streaming algorithm over mobile ad hoc networks. The proposed algorithm selects the most efficient PHY mode of IEEE 802.11 multi-rate service based on the information available at the application, MAC, and physical layers at each node in the distributed way in order to minimize end-to-end delay and remain packet loss rate in tolerable range at the final destination. Finally, experimental results are provided to show the superior performance of the proposed algorithm.

This work presents a tightly-coupled packet scheduling and routing algorithm to effectively transmit delay- sensitive media over mobile ad hoc network. First, packet urgency, node urgency, and route urgency are defined based on the end-to-end delay requirement of each packet and the number of hops over a route. Packet scheduling algorithm and packet drop policy are designed to maximize the number of packets delivered in the tolerable delay bound and minimize their node urgency at each node simultaneously. Finally, the routing algorithm is implemented to search for a route with the minimum route urgency.

This paper presents an overlay multicast tree construction algorithm for the synchronized real-time media multicast service over the Internet. The proposed tree construction algorithm is implemented to reduce not only the average delay of group members in order to support the real-time media service, but also the delay variance among group members to provide an effective synchronized service. Basically, orthogonal genetic algorithm is employed to obtain the near optimal tree among clusters with a low computational complexity since the given problem is NP-complete. Finally, experimental results are provided to show the superior performance of the proposed algorithm.

For space-time bit-interleaved coded modulation (ST-BICM) systems with iterative decoding, the overall performance is affected by the chosen mapping. In bit-error rate (BER) curves, one mapping reaches an error floor (EF) at a low signal-to-noise ratio (SNR), while other mappings result in a lower EF at a higher SNR. The constellation mappings are divided into groups where each group exhibits a distinctive BER curve. We show that the convergence abscissa of the system depends on the average total bit errors and the harmonic mean of the minimum squared Euclidean distance. In this letter, we characterize all mapping groups for ST-BICM with 8-phase-shift keying and present the optimal selection for each mapping group over independent fading channels

This paper presents metafile-based scalable caching and dynamic replacing algorithms for multiple videos over quality-of-service networks. A metafile for each video includes the cached-frame sequence, which is made by scalable caching algorithm minimizing client's buffer size and channel bandwidth under the general video traffic condition. Based on the metafiles, we propose a caching algorithm for multiple videos to effectively reduce both the buffer size and the required bandwidth and replacing algorithm to dynamically update the cached frames when user access pattern changes. Finally, experimental results are provided to compare the proposed algorithms with various existing caching and replacing algorithms.

The differentiated-services model has prevailed as a scalable approach to providing quality of service over the Internet. However, there are difficulties in providing the guaranteed service in terms of end-to-end systems since a differentiated-services network considers the quality of service of traffic aggregates due to the scalability. Furthermore, since the quality of service level may be time varying according to the network conditions in the case of the relative service model, it is more difficult to provide guaranteed end-to-end service when the path between two end systems includes several domains. Therefore, we study dynamic class selection mechanisms to guarantee the end-to-end quality of service for continuous media traffic with minimum networking cost over multiple differentiated-services domains.

In this work, we present an effective caching algorithm for a proxy server with finite storage size. Under general video traffic conditions, it is observed that the reduced amount of receiver buffer size and channel bandwidth depends on both the frame size and the frame position. To consider this fact, we propose an effective caching algorithm to select the positions of cached frames, based on the normalized buffer size. Finally, experimental results are provided to show the superior performance of the proposed algorithm.

The paper presents an online face recognition system through the Internet. Generally speaking, the amount of transmitted data is related to the time delay over the Internet, and thus a compressed image is transmitted to the remote server to reduce the time delay. First we investigate the relationships among the face recognition rate, the compression ratio and the image size, and then present an effective multi-frame-based online face recognition system based on the observations. Finally, experimental results tested on the real Internet are provided to show the superior performance of the proposed system

This paper presents a region-based video codec, which is compatible with the H.263+ standard, and its associated rate control algorithm for low variable-bit-rate (VBR) video. The proposed region-based coding scheme is a hybrid method that incorporates traditional block DCT coding as well as object-based coding. To achieve this, we adopt H.263+ as the platform, and develop a fast macroblock-based segmentation method to implement the new region-based codec. The associated rate control solution includes rate control in three levels: encoding frame selection, frame-layer rate control and macroblock-layer rate control. The goal is to improve human visual perceptual quality at low bit rates. The efficiency of the proposed rate control algorithm applied to the region-based H.263+ codec is demonstrated via several typical test sequences.

This paper presents an effective video-on-demand server architecture for the robust transmission under time-varying channel environments and traffic-smoothing/admission control/rate adaptation algorithms under RCBR network. The proposed video-on-demand server system consists of video DB and metadata DB that includes various compression history information such as the quantization parameters versus output rate profile, encoded frame positions and encoding types, etc. The traffic-smoothing, rate adaptation and admission control algorithms are performed based on these metadata.

A novel rate control algorithm with a variable-encoding frame rate method is proposed for low-bit-rate video coding. Most existing rate control algorithms for low-bit-rate video focus on bit allocation at the macroblock level under a constant frame-rate assumption. The proposed rate control algorithm is able to adjust the encoding frame rate at the expense of tolerable time-delay. The new rate control algorithm attempts to achieve a good balance between spatial quality and temporal quality to enhance the overall human perceptual quality at low-bit-rates. It is demonstrated that the rate control algorithm achieves higher coding efficiency at low-bit-rates, with a low additional computational cost. The proposed variable-encoding frame rate method is compatible with the bit-stream structure of H.263+.

A feedback-based Internet video transmission scheme based on ITU-T H.263+ is presented. The proposed system is capable of continually accommodating its stream size and managing the packet loss recovery in response to network condition changes. It consists of multiple components: TCP-friendly end-to-end congestion control and available bandwidth estimation, encoding frame-rate control and delay-based smoothing at the sender, media-aware packetization and packet loss recovery tied with congestion control, and quality recovery tools such as motion-compensated frame interpolation at the receiver. These components are designed to meet the low computational complexity requirement so that the whole system can operate in real-time. Among these, the video-aware congestion control known as a receiver-based congestion control mechanism, the variable frame-rate H.263+ encoding, and fast motion-compensated frame interpolation components are key features. Through a seamless integration, it is demonstrated that network adaptivity is enhanced enough to mitigate the packet loss and bandwidth fluctuation, resulting in a more smooth video experience at the receiver.

A region-based video codec based on the H.263+ standard is examined and its associated novel rate control schemes are proposed for visual communication through time-varying low-bit-rate channels. The region-based coding scheme is a hybrid method that consists of the traditional block DCT coding and the object-based coding. Basically, we adopt H.263+ as the platform, and develop a fast macroblock-based segmentation method to implement the region-based video codec. The proposed rate control solution includes rate control in three levels: encoding frame selection, frame-layer rate control and macroblock-layer rate control. The goal is to enhance the visual quality of decoded frames without obvious motion unsmoothness under time-varying channels. The efficiency of the proposed rate control schemes applied to the region-based video codec is demonstrated via several typical test sequences.

A novel rate control algorithm consisting of two major components, i.e. a variable encoding frame rate method and a hybrid DCT/wavelet I-frame coding scheme, is proposed for low bit rate video coding. Most existing rate control algorithms for low bit rate video focus on bit allocation at the macroblock level under a constant frame rate assumption. The proposed rate control algorithm is able to adjust the encoding frame rate at the expense of tolerable time-delay. Furthermore, an R-D optimized hybrid DCT/wavelet scheme is used for effective I-frame coding. The new rate-control algorithm attempts to achieve a good balance between spatial quality and temporal quality to enhance the overall human perceptual quality at low bit rates. It is demonstrated that the rate control algorithm achieves higher coding efficiency at low bit rates with a low additional computational cost. The variable frame rate method and the hybrid I-frame coding scheme are compatible with the bitstream structure of H.263+.

A new way to implement the Internet modem video transmission is presented. This system is capable of continually accommodating its bitstream size in response to changing network conditions. The key idea is to adopt an adaptive least mean squares (LMS) controller to orchestrates an H.263+ encoder rate control at the server end as well as a fast frame interpolation at the client end. It is demonstrated that the user datagram protocol (UDP) packet loss is significantly reduced and a smooth video display can be achieved.