An adaptive multimedia streaming dissemination system for vehicular networks

An adaptive seamless streaming dissemination system for vehicular networks is presented in this work. An adaptive streaming system is established at each local server to prefetch and buffer stream data. The adaptive streaming system computes the parts of prefetched stream data for each user and stores them temporarily at the local server, based on current situation of the users and the environments where they are located. Thus, users can download the prefetched stream data from the local servers instead of from the Internet directly, meaning that the video playing problem caused by network congestion can be avoided. Several techniques such as stream data prefetching, stream data forwarding, and adaptive dynamic decoding were utilized for enhancing the adaptability of different users and environments and achieving the best transmission efficiency. Fuzzy logic inference systems are utilized to determine if a roadside base station or a vehicle can be chosen to transfer stream data for users. Considering the uneven deployment of BSs and vehicles, a bandwidth reservation mechanism for premium users was proposed to ensure the QoS of the stream data premium users received. A series of simulations were conducted, with the experimental results verifying the effectiveness and feasibility of the proposed work.     

Server side,play buffer based quality control for adaptive media streaming

Existing media streaming protocols provide bandwidth adaptation features in order to deliver seamless video streams in an abrupt bandwidth shortage on the networks. For instance, popular HTTP streaming protocols such as HTTP Live Streaming (HLS) and MPEG-DASH are designed to select the most appropriate streaming quality based on client side bandwidth estimation. Unfortunately, controlling the quality at the client side means the effectiveness of the adaptive streaming is not controlled by service providers, and it harms the consistency in quality-of-service. In addition, recent studies show that selecting media quality based on bandwidth estimation may exhibit unstable behavior in certain network conditions. In this paper, we demonstrate that the drawbacks of existing protocols can be overcome with a server side, buffer based quality control scheme. Server side quality control solves the service quality problem by eliminating client assistance. Buffer based control scheme eliminates the side effects of bandwidth based stream selection. We achieve this without client assistance by designing a play buffer estimation algorithm. We prototyped the proposed scheme in our streaming service testbed which supports pre-transcoding and live-transcoding of the source media file. Our evaluation results show that the proposed quality control performs very well both in simulated and real environments.     

Diverse community: Demand differentiation in P2P live streaming

Peer-to-Peer (P2P) technology has become an attractive approach for enabling large-scale video streaming applications, but the factor of users’ subjective preferences is not paid enough attention in such networks. As users have various demand on video qualities, we can provide them with video streams at different resolutions without impairing their satisfaction. The adaptive streaming rate technique is a promising method. However, in providing adaptive streaming rate services, P2P live streaming design faces the following challenge: how to provide all users with uninterrupted video with their desired qualities in case that their demand dynamically changes? To shed more light on this problem, we first derive a model and formulate the problem as a resource demand vs supply problem. Then we present a framework to address the challenge via efficient bandwidth allocation and group cooperation. Through comprehensive simulations, we evaluate the effectiveness of the proposed framework, and conclude that it effectively helps existing solutions, such as Partial Participation Scheme (PPS), achieve better performance.     

Network-adaptive autonomic transcoding algorithm for seamless streaming media service of mobile clients

As a result of improvements in wireless communication technologies, a multimedia data streaming service can now be provided for mobile clients. Since mobile devices have low computing power and work on a low network bandwidth, a transcoding technology is needed to adapt the original streaming media for mobile environments. However, wireless networks have variable bandwidths depending on the movement of clients and the communication distance from Access Point (AP). These characteristics make it hard to support stable Quality of Service (QoS) streams for mobile clients. In this paper, a target transcoding bit-rate decision algorithm is proposed to provide stable QoS streams for mobile clients. In our experiments, the proposed algorithm provides seamless streaming media services based on the network adaptive bit rate control and reduces transmission failure.     

Adaptive receiver-driven approach in P2P live streaming networks

Efficient delivery under changing network conditions is a major challenge in the design of live streaming systems. This study analyzes the key considerations and factors influencing live stream quality during system operations, and attempts to improve current P2P (peer-to-peer) live streaming systems by allowing users to enjoy high quality of service under the limitations of network resources. The proposed R-D (rate-distortion) optimized dynamic nodes join algorithm is based on multipath streaming concept and receiver-driven approach. This distributed algorithm enables the system to evaluate the current network status in order to optimize the end-to-end distortion of P2P networks. Experimental results of this study demonstrate the effectiveness of the proposed approach.     

Enabling adaptive live streaming in P2P multipath networks

Live Internet streaming can be regarded as a major current multimedia delivery mode. Efficient delivery under changing network conditions is a severe challenge in the design of live streaming systems. This study analyzes the key considerations and factors influencing live stream quality during system operations, and attempts to improve present P2P (peer-to-peer) live streaming systems by allowing users to enjoy high quality of service under the limitations of network resources. The proposed R-D (Rate-Distortion) optimized-dynamic-nodes-join algorithm is based on multipath streaming concept and receiver-driven approach. This distributed algorithm enables the system to evaluate the current network status in order to optimize the end-to-end distortion of P2P networks and stay in the optimal status. Experiment results of this study demonstrate the effectiveness of the proposed approach.     

An efficient playout smoothing mechanism for layered streaming in P2P networks

Layered video streaming in peer-to-peer (P2P) networks has drawn great interest, since it can not only accommodate large numbers of users, but also handle peer heterogeneity. However, there’s still a lack of comprehensive studies on chunk scheduling for the smooth playout of layered streams in P2P networks. In these situations, a playout smoothing mechanism can be used to ensure the uniform delivery of the layered stream. This can be achieved by reducing the quality changes that the stream undergoes when adapting to changing network conditions. This paper complements previous efforts in throughput maximization and delay minimization for P2P streaming by considering the consequences of playout smoothing on the scheduling mechanisms for stream layer acquisition. The two main problems to be considered when designing a playout smoothing mechanism for P2P streaming are the fluctuation in available bandwidth between peers and the unreliability of user-contributed resources—particularly peer churn. Since the consideration of these two factors in the selection and scheduling of stream layers is crucial to maintain smooth stream playout, the main objective of our smoothing mechanism becomes the determination of how many layers to request from which peers, and in which order. In this work, we propose a playout smoothing mechanism for layered P2P streaming. The proposed mechanism relies on a novel scheduling algorithm that enables each peer to select appropriate stream layers, along with appropriate peers to provide them. In addition to playout smoothing, the presented mechanism also makes efficient use of network resources and provides high system throughput. An evaluation of the performance of the mechanism demonstrates that the proposed mechanism provides a significant improvement in the received video quality in terms of lowering the number of layer changes and useless chunks while improving bandwidth utilization.     

QoS-aware roadside base station assisted routing in vehicular networks

The transmission technology for intelligent transportation systems can be typically classified into two categories, namely, road-to-vehicle communication (RVC) and inter-vehicle communication (IVC). RVCs perform the information communication service offer from road to vehicle whereas the IVCs perform the information communication through vehicles. This work proposes quality of service (QoS)-aware roadside base station assisted routing mechanisms to establish a routing path in IVC with the assistance of roadside base station. A link failure prevention mechanism is employed to effectively construct alternative routing path required by the volatile network topology in vehicular Ad hoc networks. Besides, a bandwidth consumption predictor is presented to avoid dropping packets owing to inadequate bandwidth during handoffs. A neural network with fast learning algorithm is adopted as the core module for estimating the parameters used in the proposed schemes. Simulation results demonstrate the effectiveness and feasibility of the proposed work.     

A multi-layer experimental study of multimedia and QoS communication in wireless mesh networks

A significant issue in Mesh networks is to support multimedia transmissions while providing Quality of Service (QoS) guarantees to mobile users. For real-time multimedia streaming, unstable throughput or insufficient bandwidth will incur unexpected delay or jitter, and it remains difficult to provide comprehensive service guarantees for a wireless mesh environment. In this paper, we target the problem of providing multimedia QoS in wireless mesh networks. We design and implement a campus test-bed for supporting multimedia traffic in mobile wireless mesh networks, and investigate in detail some possible improvements on a number of layers to enable multimedia transmission over wireless networks with QoS support. We first study a number of improvements of some existing routing protocols to support multimedia transmissions. Some new admission control and rate control mechanisms are studied and their performance gains are verified in our experiments. In our new cross-layer adaptive rate control (CLARC) mechanism, we adaptively change the video encoder’s output bit rate based on the available network bandwidth to improve the quality of the received video. We also design a mobile gateway protocol to connect the MANET to Internet and a wireless LAN management protocol to automatically manage WLAN to provide some QoS.     

An efficient hybrid push-pull methodology for peer-to-peer video live streaming system on mobile broadcasting social media

With the rapid growth of wireless communication technology, the availability of highly flexible and video-friendly mobile terminal platforms (such as smartphones and tablets), the emergence of major video content providers (like YouTube, Ustream, and PPTV, which provide a large catalog of attractive contents), Peer-to-Peer (P2P) live video streaming over the wireless and Internet is becoming more and more attractive to users. One of the main challenges is to provide a good quality of service though the dynamic behavior of the network. Traditionally, tree-based model uses a push method, that broadcaster transfers data to other users. This model has low start-up delay. However, there are two main problems in this method: if the bandwidth of an internal node is low, children nodes may lose data and when an internal node failure, other nodes can’t receive data until completing the recovery of the tree. On the other hand, mesh-based model uses a pull method, has low bandwidth of a neighbor node by pulling necessary data from a number of neighbor nodes. However, mesh-based model requires large buffers to support pull data from neighbor peers and there is an adjustment between minimum delay by sending pull request and overhead of whole system. So, both models have their own strengths and weaknesses. This paper proposes a new hybrid push-pull live P2P video streaming protocol called MobileCast that combines the benefits of pull and push mechanisms for live video delivery. We present new topology for P2P network with more stable and provide better video streaming quality. Our main goal is to minimize the network end-to-end delay, startup time, overhead, packet loss compared to the pure mesh networks, pure tree networks and provide a good quality of service though the dynamic behavior of the network.

     

A token-based incentive mechanism for video streaming applications in peer-to-peer networks

Free-riding is one of the main challenges of Peer-to-Peer (P2P) streaming systems which results in reduction in video streaming quality. Therefore, providing an incentive mechanism for stimulating cooperation is one of the essential requirements to maintain video Quality of Experience (QoE) in such systems. Among the existing mechanisms, payment-based schemes are most suitable for streaming applications due to their low overhead. However, to date, no dynamic payment mechanism has been proposed which can take the stochastic dynamics of the video streaming ecosystem (e.g., the request arrival, demand submission, bandwidth availability, etc.) into account. In this paper, we propose a dynamic token-based payment mechanism in which each peer earns tokens by admitting other peers’ requests and spends tokens for submitting its demands to the others. This system allows the peers to dynamically adjust their income level in adaptation to changes in the system state. We propose a Constrained Markov Decision Process (CMDP) formulation in which the goal of each peer is to obtain a request admission policy which minimizes the expected cumulative cost of consumed bandwidth, while satisfying a long-term constraint on the Mean Opinion Score (MOS) of the users as the measure of QoE. The proposed admission policy is adaptive to the request arrival process, bandwidth state and the token bucket length of each peer. To make up for the lack of design-time knowledge of the system’s statistics, each individual peer is equipped with a model-free algorithm to learn its optimal admission policy over the course of real-time interaction with the system. Simulation results are presented to compare the performance of the proposed algorithm against baseline schemes such as: random, token-threshold, bandwidth-threshold and myopic algorithms.     

Seamless handover between unicast and multicast multimedia streams

With the deployment of heterogeneous networks, mobile users are expecting ubiquitous connectivity when using applications. For bandwidth-intensive applications such as Interact Protocol Television （IPTV）, multimedia contents are typical- ly transmitted using a multicast delivery method due to its bandwidth efficiency. However, not all networks support multicasting. Multicasting alone could lead to service disruption when the users move from a multicast-capable network to a non-multicast network. In this paper, we propose a handover scheme called application layer seamless switching （ALSS） to provide smooth real-time multimedia delivery across unicast and multicast networks. ALSS adopts a soft handover to achieve seamless playback during the handover period. A real-time streaming testbed is implemented to investigate the overall handover performance, espe- cially the overlapping period where both network interfaces are receiving audio and video packets. Both the quality of service （QoS） and objective-mapped quality of experience （QoE） metrics are measured. Experimental results show that the overlapping period takes a minimum of 56 and 4 ms for multicast-to-unicast （M2U） and unicast-to-multicast （U2M） handover, respectively. The measured peak signal-to-noise ratio （PSNR） confirms that the frame-by-frame quality of the streamed video during the handover is at least 33 dB, which is categorized as good based on ITU-T recommendations. The estimated mean opinion score （MOS） in terms of video playback smoothness is also at a satisfactory level.     

Impact of queuing discipline on packet delivery latency in ad hoc networks

Delivering live multimedia streaming over ad hoc networks can improve coordination in battlefields, assist in disaster recovery operations, and help prevent vehicular traffic accidents. However, ad hoc networks often experience congestion faster than wired networks, leading to high end-to-end delays and jitter even for moderate traffic. This paper describes a partial remedy that applies to delay sensitive but loss tolerant applications such as live streaming. We find that under relatively high UDP traffic load, the Last-In–First-Out (LIFO) with Frontdrop queuing discipline achieves less than half the delay of the commonly used First-In–First-Out (FIFO) with Taildrop, while maintaining similar jitter. In low traffic situations, FIFO and LIFO have similar delays, but FIFO with Frontdrop has the lowest jitter. The results can be applied to an adaptive queuing mechanism that changes the queuing discipline at nodes function of the locally observed traffic load. The advantage of such an approach is that it does not require new protocols and does not incur any network overhead.     

Probabilistic Adaptive Anonymous Authentication in Vehicular Networks

Vehicular networks have attracted extensive attention in recent years for their promises in improving safety and enabling other value-added services. Most previous work focuses on designing the media access and physical layer protocols. Privacy issues in vehicular systems have not been well addressed. We argue that privacy is a user-specific concept,and a good privacy protection mechanism should allow users to select the levels of privacy they wish to have. To address this requirement,we propose an adaptive anonymous authentication mechanism that can trade off the anonymity level with computational and communication overheads(resource usage) . This mechanism,to our knowledge,is the first effiort on adaptive anonymous authentication. The resources used by our protocol are few. A high traffic volume of 2000 vehicles per hour consumes about 60kbps bandwidth,which is less than one percent of the bandwidth of DSRC(Dedicated Short Range Communications) . By using adaptive anonymity,the protocol response time can further be improved 2～4 times with less than 20% bandwidth overheads.     

Optimization for adaptive bandwidth reservation in wireless multimedia networks

In future wireless multimedia networks, user mobility management for seamless connection regarding realtime multimedia applications is one of the most important problems. In this paper we propose an opportunity-cost concept-based approach for adaptive bandwidth reservation with admission control for handover calls utilizing network traffic information. Excessive reservation guarantees low blocking probability of handover calls at the cost of high blocking probability of new calls. According to our survey, however, it may degrade bandwidth utilization while no prioritization for handover admissions degrades quality of service (QoS) for ongoing calls. We consider both QoS assurance and bandwidth utilization in order to optimize the amount of bandwidth to reserve for handover admissions. We believe that our scheme could be utilized as a guideline for cost-effective radio resource allocation in mobile multimedia networks.     

基于服务质量保证的流媒体服务器设计

针对视频直播系统的特性、不稳定的Internet网络环境,重点研究了流媒体服务器服务质量保证措施,设计并实现了一种基于可扩展编码的视频直播流媒体服务器软件系统.该系统提供网络环境自适应的具有一定QoS保证的视频传输服务,已应用于国内大型通信公司.实际运行结果表明该系统提高了数据传送率,降低了网络延迟.     

Adaptive interface selection over cloud-based split-layer video streaming via multi-wireless networks

As mobile devices such as tablet PCs and smartphones proliferate, the online video consumption over a wireless network has been accelerated. From this phenomenon, there are several challenges to provide the video streaming service more efficiently and stably in the heterogeneous mobile environment. In order to guarantee the QoS of real-time HD video services, the steady and reliable wireless mesh is necessary. Furthermore, the video service providers have to maintain the QoS by provisioning streaming servers to respond the clients’ request of different video resolution. In this paper, we propose a reliable cloud-based video delivery scheme with the split-layer SVC encoding and real-time adaptive multi-interface selection over LTE and WiFi links. A split-layer video streaming can effectively scale to manage the required channels on each layer of various client connections. Moreover, split-layer SVC model brings streaming service providers a remarkable opportunity to stream video over multiple interfaces (e.g. WiFi, LTE, etc.) with a separate controlling based on their network status. Through the adaptive interface selection, the proposed system aims to ensure the maximizing video quality which the bandwidth of LTE/WiFi accommodates. In addition, the system offers cost-effective streaming to mobile clients by saving the LTE data consumption. In our system, an adaptive interface selection is developed with two different algorithms, such as INSTANT and EWMA methods. We implemented a prototype of mobile client based on iOS particularly by using iPhone5S. Moreover, we also employ the split-layer SVC encodes in streaming server-side as the add-on module to SVC reference encoding tool in a virtualized environment of KVM hypervisor. We evaluated the proposed system in an emulated and a real-world heterogeneous wireless network environments. The results show that the proposed system not only achieves to guarantee the highest quality of video frames via WiFi and LTE simultaneous connection, but also efficiently saves LTE bandwidth consumption for cost-effectiveness to client-side. Our proposed method provides the highest video quality without deadline misses, while it consumes 50.6% LTE bandwidth of ‘LTE-only’ method and 72.8% of the conventional (non-split) SVC streaming over a real-world mobile environment.     

Modeling and evaluating IPTV applications in WiMAX networks

Internet Protocol-based Television (IPTV) is a digital television service which delivers television content via an IP network. The rapid growth of wireless network technology in recent years has changed, the way people access the Internet. Adding mobility to IPTV can create a truly compelling ubiquitous service which spans different network domains and varied IP-enabled terminals and devices, such as set-top boxes, PCs and cell phones. However, extending IPTV service to wireless networks requires overcoming bandwidth bottlenecks and high packet loss rates. Following the IEEE 802.16 standard, worldwide interoperability for microwave access (WiMAX) features high data rates and large service coverage, offering a wireless broadband solution for IPTV services. While previous research has focused on creating a broadband IPTV service few studies have practically evaluated IPTV applications in a wireless broadband network environment. In this paper, we model and evaluate a common constant bit rate (CBR)¹ based IPTV application and an IPTV live streaming (PPStreaming)² application while retrieving IPTV content via a WiMAX network. We also use the NS2 simulation tool to evaluate the performance of these two IPTV applications. The evaluation metrics include latency, packet loss, data rate and throughput statistics when the two IPTV applications are run in the WiMAX network. ¹The simplest IPTV solution is to convey video content by CBR. IPTV operators and content delivery networks relay CBR streaming content to control the demand for network capacity. Broadcasters prefer CBR video as it conserves bandwidth resources, but CBR delivery can degrade video quality result in higher overall demand on network capacity. ²PPStreaming (also referred to as P2P streaming Internet TV) is a network for live media streaming. In principle it’s similar to BitTorrent (BT) in that it provides stable and smooth broadcast of TV programs to broadband users. Unlike traditional streaming media, PPStreaming adopts P2P-streaming technology and supports full-scale visits with tens of thousands of simultaneous users. Its client software can be used in the browser or as a standalone executable.     

A rate-based flow control mechanism for avoiding congestion

The rate-based flow control mechanisms for the Available Bit Rate(ABR)service are used to share the available badwidth of a bottleneck switch connected to a bottleneck link fairly and reasonably among many competitive users,and to maintain the buffer queue length of the witch at a desired level in order to avoid congestion in Asynchronous Transfer Mode(ATM)networks.In this Paper,a control theoretic approach that uses a Deadbeat-Response(DR) controller to the desing of a rate-based flow control mechanism is presented.The mehanism has a simple structure and is robust in the sense that its stability is not sensitive to the change of the number of active Virtual Connections(VCs),Simulation results show that this mechanism not only ensures fair share of the bandwidth for all active VCs regardless of the nmuber of hops they traverse but also has the advantages of fast convergence ,no oscillation,and high link bandwidth utilization.     

Utility-oriented resource allocation for 360-degree video transmission over heterogeneous networks

Immersive media streaming, especially virtual reality (VR)/360-degree video streaming which is very bandwidth demanding, has become more and more popular due to the rapid growth of the multimedia and networking deployments. To better explore the usage of resource and achieve better video quality perceived by users, this paper develops an application-layer scheme to jointly exploit the available bandwidth from the LTE and Wi-Fi networks in 360-degree video streaming. This newly proposed scheme explores the saliency of video contents, prediction of users' view and channel status information to maximize the system's utility in a multi-RAN environment. It also determines the optimal association of users with multiple Wi-Fi access points (APs). Besides, a novel buffer strategy is proposed to mitigate the influence of short-time prediction problem for transmitting 360-degree videos in time-varying networks. The promising performance and low complexity of the proposed scheme and algorithms are validated in simulations with various 360-degree videos.