Efficient scalable video multicast based on network-coded communication

In order to increase the efficiency of mobile video transmission in a 5G network, this paper investigates a cooperative multicast of scalable video using network coding with adaptive modulation and coding over dedicated relay-based cellular networks. Different scalable video layers prefer different protection degrees, and user equipments (UEs) in different locations experience different packet loss rates in wireless networks. Guaranteeing that all UEs experience a certain level of video quality is one of the biggest challenges in scalable video multicast. Using the number of satisfied UEs as a metric, the proposed efficient scalable video multicast based on network-coded cooperation (SVM-NC) scheme, combined with adaptive modulation and coding, enhances the attainable system performance under strict time and bandwidth resource constraints for guaranteed smooth playback. Various simulations were performed for performance evaluation. The proposed scheme ensures that the expected percentage of satisfied UEs approximately achieves the maximum number of UEs in a multicast group by using network-coded cooperation over dedicated relay-based cellular networks. In addition, the peak signal-to-noise ratio metric is asymptotic to the maximum performance of high-resolution video quality offered by service providers.     

Design and Implementation of a Network‐Adaptive Mechanism for HTTP Video Streaming

This paper proposes a network‐adaptive mechanism for HTTP‐based video streaming over wireless/mobile networks. To provide adaptive video streaming over wireless/mobile networks, the proposed mechanism consists of a throughput estimation scheme in the time‐variant wireless network environment and a video rate selection algorithm used to increase the streaming quality. The adaptive video streaming system with proposed modules is implemented using an open source multimedia framework and is validated over emulated wireless/mobile networks. The emulator helps to model and emulate network conditions based on data collected from actual experiments. The experiment results show that the proposed mechanism provides higher video quality than the existing system provides and a rate of video streaming almost void of freezing.     

A handover-aware seamless video streaming scheme in heterogeneous wireless networks

With the development of wireless technologies, video streaming services over heterogeneous wireless networks have become more popular in recent years. Video streaming schemes for heterogeneous networks should consider vertical handover in which the link capacity is varied significantly, because the quality experienced for a video streaming service is affected by the network status. When a vertical handover occurs, an abrupt bandwidth change and substantial handover latency lead to bursty packet loss and discontinuity of the video playback. In this paper, we propose a handover-aware video streaming scheme to provide seamless video streaming services over heterogeneous wireless networks. The proposed scheme adjusts its sending rate and the quality level of the transmitted video streams according to the significant bandwidth variation that occurs in a vertical handover. To expedite the response to the bandwidth variation due to a handover, our scheme uses an explicit notification message that informs the streaming server of a client's handover occurrence. In order to evaluate the performance, we use a simulation environment for a vertical handover between wireless local area networks and cellular networks. Through the simulation results, we prove that our scheme improves the experienced quality of video streaming in vertical handovers.     

基于IIS平滑流式处理的移动流媒体系统

针对当前流媒体传输协议在无线网络中的不足,采用IIS平滑流式处理技术设计了基于服务器/客户端模式的移动流媒体系统。详细讨地论了微软的IIS平滑流式处理协议,采用该协议传输音/视频数据,搭建了基于IIS7 Web服务器的流媒体服务,设计了一款基于Windows Phone 7智能手机的流媒体播放器客户端。通过在WiFi网络环境下测试该系统,在直播和点播两种播放模式下,客户端播放的视频画面清晰流畅。通过仿真无线网络带宽的变化,验证了IIS平滑流式处理根据客户端的可用带宽实时调整传送到客户端视频流的质量的特性。     

End-to-end optimized TCP-friendly rate control for real-time video streaming over wireless multi-hop networks

Rate control is an important issue in video streaming applications. The most popular rate control scheme over wired networks is TCP-Friendly Rate Control (TFRC), which is designed to provide optimal transport service for unicast multimedia delivery based on the TCP Reno’s throughput equation. It assumes perfect link quality, treating network congestion as the only reason for packet losses. Therefore, when used in wireless environment, it suffers significant performance degradation because of packet losses arising from time-varying link quality. Most current research focuses on enhancing the TFRC protocol itself, ignoring the tightly coupled relation between the transport layer and other network layers. In this paper, we propose a new approach to address this problem, integrating TFRC with the application layer and the physical layer to form a holistic design for real-time video streaming over wireless multi-hop networks. The proposed approach can achieve the best user-perceived video quality by jointly optimizing system parameters residing in different network layers, including real-time video coding parameters at the application layer, packet sending rate at the transport layer, and modulation and coding scheme at the physical layer. The problem is formulated and solved as to find the optimal combination of parameters to minimize the end-to-end expected video distortion constrained by a given video playback delay, or to minimize the video playback delay constrained by a given end-to-end video distortion. Experimental results have validated 2–4 dB PSNR performance gain of the proposed approach in wireless multi-hop networks by using H.264/AVC and NS-2.     

QoE in video streaming over wireless networks: perspectives and research challenges

The deployment of 3G/LTE networks and advancements in smart mobile devices had led to high demand for multimedia streaming over wireless network. The rapid increasing demand for multimedia content poses challenges for all parties in a multimedia streaming system, namely, content providers, wireless network service providers, and smart device makers. Content providers and mobile network service providers are both striving to improve their streaming services while utilizing advancing technologies. Smart device makers endeavor to improve processing power and displays for better viewing experience. Ultimately, the common goal shared by content providers, network service providers, and smart device manufactures is to improve the QoE for users. QoE is both an objective and a subjective metric measuring the streaming quality experience by end users. It may be measured by streaming bitrate, playback smoothness, video quality metrics like Peak to Signal Noise Ratio, and other user satisfaction factors. There have been efforts made to improve the streaming experiences in all these aspects. In this paper, we conducted a survey on existing literatures on QoE of video streaming to gain a deeper and more complete understanding of QoE quality metrics. The goal is to inspire new research directions in defining better QoE and improving QoE in existing and new streaming services such as adaptive streaming and 3D video streaming.     

A novel adaptive logic for dynamic adaptive streaming over HTTP

In this paper, we propose an estimation method that estimates the throughput of upcoming video segments based on variations in the network throughput observed during the download of previous video segments. Then, we propose a rate-adaptive algorithm for Hypertext Transfer Protocol (HTTP) streaming. The proposed algorithm selects the quality of the video based on the estimated throughput and playback buffer occupancy. The proposed method selects high-quality video segments, while minimizing video quality changes and the risk of playback interruption, improving user’s experience. We evaluate the algorithm for single- and multi-user environments and demonstrate that it performs remarkably well under varying network conditions. Furthermore, we determine that it efficiently utilizes network resources to achieve a high video rate; competing HTTP clients achieve equitable video rates. We also confirm that variations in the playback buffer size and segment duration do not affect the performance of the proposed algorithm.     

移动网络中流媒体业务适配承载带宽的解决方案研究

YouTube、优酷、PPLive等一大批以流媒体为载体的视频网站的火爆,意味着流媒体时代的到来。然而,由于移动网络通信环境的不确定性,不能提供稳定的带宽,不能保障流媒体的业务质量。在研究流媒体业务特征、通信标准的基础上,提出一种基于移动核心网信令的,向流媒体业务平台告知移动网络无线链路实时带宽的方法。流媒体平台可根据当前带宽,及时地调整流媒体编码速率,以达到节约网络资源,改善用户业务感受的效果。     

Client intelligence for adaptive streaming solutions

In state-of-the-art adaptive streaming solutions, to cope with varying network conditions, the client side can switch between several video copies encoded at different bit-rates during streaming. Each video copy is divided into chunks of equal duration. To achieve continuous video playback, each chunk needs to arrive at the client before its playback deadline. The perceptual quality of a chunk increases with the chunk size in bits, whereas bigger chunks require more transmission time and, as a result, have a higher risk of missing transmission deadline. Therefore, there is a trade-off between the overall video quality and continuous playback, which can be optimized by proper selection of the next chunk from the encoded versions. This paper proposes a method to compute a set of optimal client strategies for this purpose.     

A distributed utility‐based scheduling for peer‐to‐peer video streaming over wireless networks

Interactive multimedia applications such as peer‐to‐peer (P2P) video services over the Internet have gained increasing popularity during the past few years. However, the adopted Internet‐based P2P overlay network architecture hides the underlying network topology, assuming that channel quality is always in perfect condition. Because of the time‐varying nature of wireless channels, this hardly meets the user‐perceived video quality requirement when used in wireless environments. Considering the tightly coupled relationship between P2P overlay networks and the underlying networks, we propose a distributed utility‐based scheduling algorithm on the basis of a quality‐driven cross‐layer design framework to jointly optimize the parameters of different network layers to achieve highly improved video quality for P2P video streaming services in wireless networks. In this paper, the quality‐driven P2P scheduling algorithm is formulated into a distributed utility‐based distortion‐delay optimization problem, where the expected video distortion is minimized under the constraint of a given packet playback deadline to select the optimal combination of system parameters residing in different network layers. Specifically, encoding behaviors, network congestion, Automatic Repeat Request/Query (ARQ), and modulation and coding are jointly considered. Then, we provide the algorithmic solution to the formulated problem. The distributed optimization running on each peer node adopted in the proposed scheduling algorithm greatly reduces the computational intensity. Extensive experimental results also demonstrate 4–14 dB quality enhancement in terms of peak signal‐to‐noise ratio by using the proposed scheduling algorithm. Copyright © 2015 John Wiley & Sons, Ltd.     

Cross-Layer Optimized Rate Adaptation and Scheduling for Multiple-User Wireless Video Streaming

We present a cross-layer optimized video rate adaptation and user scheduling scheme for multi-user wireless video streaming aiming for maximum quality of service (QoS) for each user,, maximum system video throughput, and QoS fairness among users. These objectives are jointly optimized using a multi-objective optimization (MOO) framework that aims to serve the user with the least remaining playback time, highest delivered video seconds per transmission slot and maximum video quality. Experiments with the IS-856 (1timesEV-DO) standard numerology and ITU pedestrian A and vehicular B environments show significant improvements over the state-of- the-art wireless schedulers in terms of user QoS, QoS fairness, and the system throughput.     

Android系统中RTMP流媒体直播的设计与实现

在人烟稀少、边远山区和高原等地区,无线基站覆盖面较小导致无线传输通道环境恶劣,无法有效实现远程高清音视频的实时播放.前端无线手持设备多采用Android操作系统,因此Android系统高质量实时传输高清音视频成为远程视频监控的关键问题.为保证高清音视频的稳定传输,解决网络拥塞和画面延迟抖动等问题,在Android系统上采用RTMP协议传输高清音视频,优化RTMP的数据传输策略,实现了高清音视频的无线传输.在严重丢失帧的恶劣环境下测试该系统,系统运行稳定,显著提高了音视频播放的流畅性和传输的服务质量.     

Performance modeling of FEC‐based unequal error protection for H.264/AVC video streaming over burst‐loss channels

Unequal error protection systems are a popular technique for video streaming. Forward error correction (FEC) is one of error control techniques to improve the quality of video streaming over lossy channels. Moreover, frame‐level FEC techniques have been proposed for video streaming because of different priority video frames within the transmission rate constraint on a Bernoulli channel. However, various communication and storage systems are likely corrupted by bursts of noise in the current wireless behavior. If the burst losses go beyond the protection capacity of FEC, the efficacy of FEC can be degraded. Therefore, our proposed model allows an assessment of the perceived quality of H.264/AVC video streaming over bursty channels, and is validated by simulation experiments on the NS‐2 network simulator at a given estimate of the packet loss ratio and average burst length. The results suggest a useful reference in designing the FEC scheme for video applications, and as the video coding and channel parameters are given, the proposed model can provide a more accurate evaluation tool for video streaming over bursty channels and help to evaluate the impact of FEC performance on different burst‐loss parameters. Copyright © 2014 John Wiley & Sons, Ltd.     

End-to-end TCP-friendly streaming protocol and bit allocation for scalable video over wireless Internet

With the convergence of wired-line Internet and mobile wireless networks, as well as the tremendous demand on video applications in mobile wireless Internet, it is essential to an design effective video streaming protocol and resource allocation scheme for video delivery over wireless Internet. Taking both network conditions in the Internet and wireless networks into account, in this paper, we first propose an end-to-end transmission control protocol (TCP)-friendly multimedia streaming protocol for wireless Internet, namely WMSTFP, where only the last hop is wireless. WMSTFP can effectively differentiate erroneous packet losses from congestive losses and filter out the abnormal round-trip time values caused by the highly varying wireless environment. As a result, WMSTFP can achieve higher throughput in wireless Internet and can perform rate adjustment in a smooth and TCP-friendly manner. Based upon WMSTFP, we then propose a novel loss pattern differentiated bit allocation scheme, while applying unequal loss protection for scalable video streaming over wireless Internet. Specifically, a rate-distortion-based bit allocation scheme which considers both the wired and the wireless network status is proposed to minimize the expected end-to-end distortion. The global optimal solution for the bit allocation scheme is obtained by a local search algorithm taking the characteristics of the progressive fine granularity scalable video into account. Analytical and simulation results demonstrate the effectiveness of our proposed schemes.     

SSIM-based error-resilient rate-distortion optimization of H.264/AVC video coding for wireless streaming

The SSIM-based rate-distortion optimization (RDO) has been verified to be an effective tool for H.264/AVC to promote the perceptual video coding performance. However, the current SSIM-based RDO is not efficient for improving the perceptual quality of the video streaming application over the error-prone network, because it does not consider the transmission induced distortion in the encoding process. In this paper, a SSIM-based error-resilient RDO scheme for H.264/AVC is proposed to improve the wireless video streaming performance. Firstly, with the help of the SSE-based RDO, we present a low-complexity Lagrange multiplier decision method for the SSIM-based RDO video coding in the error-free environment. Then, the SSIM-based decoding distortion of the user end is estimated at the encoder and is correspondingly introduced into the RDO to involve the transmission induced distortion into the encoding process. Further, the Lagrange multiplier is theoretically derived to optimize the encoding mode selection in the error-resilient RDO process. Experimental results show that the proposed SSIM-based error-resilient RDO can obtain superior perceptual video quality (more structural information) to the traditional SSE-based error-resilient RDO for wireless video streaming at the same bit rate condition.     

一种用于H.264视频容错转码的帧内宏块刷新算法

无线网络信道具有误码率高、延时抖动等不稳定的特性。为了保证视频信息从有线网络到无线网络中传输的质量,需要对视频进行容错转码。基于H.264容错转码系统,利用转码结构中解码部分的运动矢量信息,自适应地在编码部分进行帧内刷新,以提高视频的抗误码能力。实验结果对比表明,该帧内刷新算法能够以较少的比特率为代价,有效地提高了接收端的视频质量。     

Energy efficient fuzzy-based DASH adaptation algorithm

With the expansive demand for video streaming over mobile networks, it is necessary to adopt schemes that balance the need for high video quality with the available network resources when streaming or downloading the video. Several approaches were proposed in the literature, including Dynamic Adaptive Streaming over HTTP (DASH). In this work, we consider an approach in which we place sufficient emphasis on the constrained battery resources in mobile devices when making decisions on the quality (or bitrate) of the video to be requested. This is done by using a fuzzy logic controller that enhances the performance of the Fuzzy-based DASH (FDASH) scheme. Simulation results show that our proposed approach conserves more energy than its predecessor while maintaining similar video quality and avoiding playback interruptions.     

Slack time–based scheduling scheme for live video streaming in P2P network

Nowadays, peer‐to‐peer network plays a significant role in data transfer and communication. The past few years have witnessed considerable growth in this area because of its inherent advantages. Peer‐to‐peer live streaming has a significant impact on video transmission over the Internet. Major factors that influence the performance of P2P live streaming are overlay construction and scheduling strategies. Although, a large number of scheduling schemes are developed but none of them is comprehensive enough to provide solutions to live streaming issues. These suffer from substantial delay and low video quality at the receiver side. In this paper, a new start‐up–based selection procedure and slack time–based scheduling scheme is proposed. The start‐up selection procedure defines the start‐up buffer location for new peer, and the scheduling scheme selects both the chunk and peers. The proposed scheduling scheme uses both push and pull priority–based strategies. The simulation results of the proposed approach demonstrates significant improvement in both the network performance and video quality at the receiver side. It is observed that playback delay, startup delay, and end‐to‐end delay in the network are reduced and quality of the video at receiver side is improved as the distortion and frame loss ratio is decreased.     

Efficient handoff based on link quality prediction for video streaming in urban transport systems

The continuously increasing interest in developing efficient vehicle‐roadside communication networks to provide on‐board connectivity has recently brought to the definition of several solutions based on different wireless technologies. Among them, wireless local area network‐based solutions emerged as an attracting alternative to guarantee high‐quality connectivity enabling services, like video streaming, that require stringent quality of service guarantees. In this work, we propose a handoff procedure based on a forecasting model of the link quality for mobile routers operating in vehicle‐roadside wireless local area network‐based networks. First, a preliminary set of experiments is performed in a realistic environment to study the behavior of the wireless channel when mobility in urban environment is considered. Then, considering the hands‐on experience gained from the initial set of experiments, a novel handoff procedure is designed, which exploits a forecasting technique to predict link channel quality. The proposed procedure is then exploited in a cross‐layer manner to proactively reduce the number of transmitted layers during handoff in the case of real‐time video traffic based on H.264/SVC encoding. The proposal is assessed by means of simulation and compared with existing solutions. Results demonstrate that our proposal guarantees performance comparable with other algorithms. The advantage of predicting the handoff point is demonstrated by means of simulations employing a realistic video streaming traffic model, showing how the quality of experience perceived by end‐users can be improved through the adaptation of the traffic load. Copyright © 2016 John Wiley & Sons, Ltd.     

基于SIP无线视频监控系统实时视频的实现

由于无线视频监控系统朝着网络化、一体化方向发展,提出了无线视频监控系统的实时视频的请求建立连接过程,采用简单性、扩展性较高的SIP作为呼叫连接控制协议,而在流媒体的传输过程中采用RTP/RTCP协议,增强实时视频灵活性,解决了实时视频对传输质量、传输速率等问题,并简要分析了客户端与服务器端的实时视频传输过程,结果显示视频图像清晰。