End-to-End QoS for Video Delivery Over Wireless Internet

Providing end-to-end quality of service (QoS) support is essential for video delivery over the next-generation wireless Internet. We address several key elements in the end-to-end QoS support, including scalable video representation, network-aware end system, and network QoS provisioning. There are generally two approaches in QoS support: the network-centric and the end-system centric solutions. The fundamental problem in a network-centric solution is how to map QoS criterion at different layers respectively, and optimize total quality across these layers. We first present the general framework of a cross-layer network-centric solution, and then describe the recent advances in network modeling, QoS mapping, and QoS adaptation. The key targets in end-system centric approach are network adaptation and media adaptation. We present a general framework of the end-system centric solution and investigate the recent developments. Specifically, for network adaptation, we review the available bandwidth estimation and efficient video transport protocol; for media adaptation , we describe the advances in error control, power control, and corresponding bit allocation. Finally, we highlight several advanced research directions.     

Adaptive cross-layer protection strategies for robust scalable video transmission over 802.11 WLANs

Robust streaming of video over 802.11 wireless local area networks poses many challenges, including coping with bandwidth variations, data losses, and heterogeneity of the receivers. Currently, each network layer (including physical layer, media access control (MAC), transport, and application layers) provides a separate solution to these challenges by providing its own optimized adaptation and protection mechanisms. However, this layered strategy does not always result in an optimal overall performance for the transmission of video. Moreover, certain protection strategies can be implemented simultaneously in several layers and, hence, the optimal choices from the application and complexity perspective need to be identified. In this paper, we evaluate different error control and adaptation mechanisms available in the different layers for robust transmission of video, namely MAC retransmission strategy, application-layer forward error correction, bandwidth-adaptive compression using scalable coding, and adaptive packetization strategies. Subsequently, we propose a novel adaptive cross-layer protection strategy for enhancing the robustness and efficiency of scalable video transmission by performing tradeoffs between throughput, reliability, and delay depending on the channel conditions and application requirements. The results obtained using the proposed adaptive cross-layer protection strategies show a significantly improved visual performance for the transmitted video over a variety of channel conditions.     

基于LTE资源块感知的自适应无线流媒体系统

自适应流媒体技术用户体验的好坏,很大程度上由流媒体应用对端到端带宽估计的准确度决定,而在高度动态的LTE网络中,对带宽的估计极具挑战。设计并实现了一种基于LTE资源块感知的自适应无线流媒体系统,系统中的LTE资源块监测模块监测当前蜂窝小区范围内的物理层资源占用情况;速率映射机制据此将视频分片下载速率映射为当前可用带宽;码率自适应算法结合当前LTE网络中潜在可用带宽和当前视频缓存状况,选择最合适的码率版本,以实现视频质量和播放卡顿之间的折中。在原型系统上的测试实验中,与两种基线码率自适应算法进行对比,所提方法在保持极低的播放卡顿率的情况下获得了最高的平均视频码率,有效提升了用户的视频观看体验。     

QoE-aware video cooperative caching and transmission mechanism based on TWDM-PON and C-RAN

The quality of user experience suffers from performance deterioration dramatically due to the explosively growing data traffic.To improve the poor performance of cell-edge users and heavy-load cell users,which caused by dense network and load imbalance respectively,an QoE-aware video cooperative caching and transmission mechanism in cloud radio access network was proposed.Cooperative gain-aware virtual passive optical network was established to provide cooperative caching and transmission for video streaming by adopting collaborative approach in optical domain and wireless domain.Furthermore,user experience for video streaming,bandwidth provisioning and caching strategy were jointly optimized to improve QoE,which utilized the methods of dynamic caching in optical domain and buffer level-aware bandwidth configuration in wireless domain.The results show that the proposed mechanism enhances the quality of user experience and effectively improves the cache hit rate.     

QoE optimization for HTTP adaptive streaming: Performance evaluation of MEC-assisted and client-based methods

Seamless streaming of high quality video under unstable network condition is a big challenge. HTTP adaptive streaming (HAS) provides a solution that adapts the video quality according to the network conditions. Traditionally, HAS algorithm runs at the client side while the clients are unaware of bottlenecks in the radio channel and competing clients. The traditional adaptation strategies do not explicitly coordinate between the clients, servers, and cellular networks. The lack of coordination has been shown to lead to suboptimal user experience. As a response, multi-access edge computing (MEC)-assisted adaptation techniques emerged to take advantage of computing and content storage capabilities in mobile networks. In this study, we investigate the performance of both MEC-assisted and client-side adaptation methods in a multi-client cellular environment. Evaluation and comparison are performed in terms of not only the video rate and dynamics of the playback buffer but also the fairness and bandwidth utilization. We conduct extensive experiments to evaluate the algorithms under varying client, server, dataset, and network settings. Results demonstrate that the MEC-assisted algorithms improve fairness and bandwidth utilization compared to the client-based algorithms for most settings. They also reveal that the buffer-based algorithms achieve significant quality of experience; however, these algorithms perform poorly compared with throughput-based algorithms in protecting the playback buffer under rapidly varying bandwidth fluctuations. In addition, we observe that the preparation of the representation sets affects the performance of the algorithms, as does the playback buffer size and segment duration. Finally, we provide suggestions based on the behaviors of the algorithms in a multi-client environment.     

一种基于罚因子的DASH调度算法

随着移动互联网的普及,基于DASH的流媒体传输协议的应用越来越广泛。如何在带宽波动较大的移动互联网环境中保证用户实现流媒体的流畅点播,提高用户的体验度是DASH调度算法最主要研究的问题。以提高用户体验度为出发点,结合带宽和缓存深度两方面因素,对带宽预测模型的置信度进行评价。在高置信度情况下,大胆地对网络带宽估计模型的模型参量进行调整;在低置信度情况下,以保护缓冲区深度为目的,谨慎地对模型参量进行调整。这种调整势必会对QoE造成相应的影响,该影响作为"罚因子"反馈回模型置信度的评价,以获得模型参数的动态最优解,得到一种较好的DASH调度算法。     

Segment‐aware dynamic routing for DASH flows over software‐defined networks

Most of the video streaming applications running over the Internet send video data over HTTP and provide an architecture for video clients to adapt video quality during streaming. In HTTP adaptive streaming, a raw video is encoded at various qualities, each encoded video file is divided into small segments, and the clients may change the segment quality by sending requests for segments having different qualities over time. MPEG has standardized dynamic adaptive streaming over HTTP (MPEG‐DASH) due to this tendency. In this work, we focus on DASH over software‐defined networks (SDN), and we dynamically reroute DASH flows by considering the current network capacity, available bandwidth of the paths, and bitrate of the segments in order to provide high quality of experience (QoE) and fairness among DASH clients. Simulations performed under various network conditions show that the proposed study provides higher QoE and fairness compared with the max‐flow routing approach.     

基于无线视频服务器自适应带宽算法

针对无线网络上行带宽有限的情况,提出了无线视频传输带宽的自适应算法。采用双卡发送采集的视频流数据,这样大大增加了无线视频传输带宽。解决了公共无线网络带宽资源有限的问题。使得无线视频传输码率能够达到500～900 kbit/s。在接收端采取双缓冲区的设计,在客户端能够得到清晰、流畅的视频图像。从而解决了无线视频传输和带宽不足的问题。     

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.     

Rate adaptation for dynamic adaptive streaming over HTTP in content distribution network

Recently the 3rd Generation Partnership Project (3GPP) and the Moving Picture Experts Group (MPEG) specified Dynamic Adaptive Streaming over HTTP (DASH) to cope with the shortages in progressive HTTP based downloading and Real-time Transport Protocol (RTP) over the User Datagram Protocol (UDP), shortly RTP/UDP, based streaming. This paper investigates rate adaptation for the serial segment fetching method and the parallel segment fetching method in Content Distribution Network (CDN). The serial segment fetching method requests and receives segments sequentially whereas the parallel segment fetching method requests media segments in parallel. First, a novel rate adaptation metric is presented in this paper, which is the ratio of the expected segment fetch time (ESFT) and the measured segment fetch time to detect network congestion and spare network capacity quickly. ESFT represents the optimum segment fetch time determined by the media segment duration multiplied by the number of parallel HTTP threads to deliver media segments and the remaining duration to fetch the next segment to keep a certain amount of media time in the client buffer. Second, two novel rate adaptation algorithms are proposed for the serial and the parallel segment fetching methods, respectively, based on the proposed rate adaptation metric. The proposed rate adaptation algorithms use a step-wise switch-up and a multi-step switch-down strategy upon detecting the spare networks capacity and congestion with the proposed rate adaptation metric. To provide a good convergence in the representation level for DASH in CDN, a sliding window is used to measure the latest multiple rate adaptation metrics to determine switch-up. To decide switch-down, a rate adaptation metric is used. Each rate adaptation metric represents a reception of a segment/portion of a segment, which can be fetched from the different edge servers in CDN, hence it can be used to estimate the corresponding edge server bandwidth. To avoid buffer overflow due to a slight mismatch in the optimum representation level and bandwidth, an idling method is used to idle a given duration before sending the next segment. In order to solve the fairness between different clients who compete for bandwidth, the prioritized optimum segment fetch time is assigned to the newly joined clients. The proposed rate adaptation method does not require any transport layer information, which is not available at the application layer without cross layer communication. Simulation results show that the proposed rate adaptation algorithms for the serial and the parallel segment fetching methods quickly adapt the media bitrate to match the end-to-end network capacity, provide an advanced convergence and fairness between different clients and also effectively control buffer underflow and overflow for DASH in CDN. The reported simulation results demonstrate that the parallel rate adaptation outperforms the serial DASH rate adaptation algorithm with respect to achievable media bitrates while the serial rate adaptation is superior to the parallel DASH with respect to the convergence and buffer underflow frequency.     

A fast algorithm of bitstream extraction using distortion prediction based on simulated annealing

Scalable video streams can be extracted to meet the bandwidth limitation of different networks and end-users. Bitstream extraction is usually performed at the network proxy or gateway during transmission, where a low computational complexity is always preferred. How to quickly and accurately select the best resolution combination for a video to meet different bandwidth requirements by each user is crucial in bitstream extraction. In this paper a fast algorithm of bitstream extraction for scalable video is proposed. The interlayer dependency between the base quality layer and the first quality layer was used to predict the distortion of higher quality layers. When quality of every layer is available, the proposed method searches for the optimized combination of quality layers based on simulated annealing. Experimental results show that the proposed method provides an optimized performance, which is significantly higher than that can be achieved by the basic extraction method. Compared to the quality layer based extraction method in the reference software model of H.264/SVC (i.e., JSVM), the proposed algorithm can greatly decrease the decoding times from 2NT to only 2 without losing rate-distortion performance. Furthermore, the proposed method obtains a more smoothed video quality which is always favorable to the observer.     

CAR: a low latency video-on-demand broadcasting scheme for heterogeneous receivers

As different types of wireless networks are converging into an all-IP network, i.e., the Internet, it can be expected that in the near future video-on-demand (VoD) will be widely applied to many interesting services, and users can access these services using heterogeneous terminals via heterogeneous wired/wireless access networks. Many periodic broadcasting protocols have been proposed to reduce the implementation cost of VoD systems. However, most of the protocols assumed homogeneity for user terminals, while in practice, user terminals are usually quite different in their processing power, buffer space, and power. To address this problem, a few periodic broadcasting protocols providing the same video quality for all heterogeneous clients have been proposed recently. In this paper, we proposed a novel heterogeneous VoD broadcasting technique called Catch and Rest (CAR) to accommodate bandwidth heterogeneity without sacrificing user video quality. Then, we provide mathematic analysis to calculate the client bandwidth and buffer space requirements of CAR. Finally, we present our performance evaluation results for CAR. Our results show that under the same system resources (i.e., server and network bandwidth), CAR provides more uniform and acceptable service latency for all heterogeneous clients compared to previous works.     

Combining the rate adaptation and quality adaptation schemes for wireless video streaming

Video streaming service over wireless networks is a challenging task because of the changes in the wireless channel conditions that can occur due to interference, fading, and station mobility. Moreover, the IEEE 802.11 WLAN standard does not contain any specifications for the rate adaptation scheme which are useful for improving the wireless link utilization. To provide efficient wireless video streaming service, the rate adaptation scheme should be applied at the low layer and the quality adaptation scheme should be considered at the high layer. To meet this requirement of wireless video streaming, we propose a new cross-layer design for video streaming over wireless networks. This design includes the rate adaptation scheme in the data link and physical layers and the quality adaptation scheme in the application layer. The rate adaptation scheme adjusts the data transmission rate based on the measured RSSI at the sender-side and informs the quality adaptation scheme about the rate limits. Then the quality adaptation scheme utilizes this rate limits to adjust the quality of the video stream. Through performance evaluations, we prove that our cross-layer design improves the wireless link utilization and the quality of the video stream simultaneously.     

Long-term rate control for concurrent multipath real-time video transmission in heterogeneous wireless networks

Concurrent multipath transmission provides an effective solution for streaming high-quality mobile videos in heterogeneous wireless networks. Rate control is commonly adopted in multimedia communication systems to fully utilize the available network bandwidth. This paper proposes a novel rate control for concurrent multipath video transmission. The existing rate control algorithms mainly adapt bit rate in the short-term pattern, i.e., without considering the long-term video transmission quality. We propose a long-term rate control scheme that takes into account the status of both the transmission buffer and video frames. First, a mathematical model is developed to formulate the non-convex problem of long-term quality maximization. Second, we develop a dynamic programming solution for online encoding bit rate control based on buffer status. The performance evaluation is conducted in a real test bed over LTE and Wi-Fi networks. Experimental results demonstrate that the proposed long-term rate control scheme achieves appreciable improvements over the short-term rate control schemes in terms of video quality and delay performance.     

Drift controlled scalable wavelet based video coding in the overcomplete discrete wavelet transform domain

Traditional video coders use the previous frame to perform motion estimation and compensation. Though they are less complex and have minimum coding delays, these coders lose their efficiency when subjected to scalability requirements. Recent 3D wavelet coders using lifting schemes offer high compression efficiency and scalability without significant loss in performance. The main drawback of 3D coders is that they process several frames at a time. This introduces additional delay, which makes them less suitable for real time applications.In this work, we propose a novel scheme to minimize drift in scalable wavelet based video coding, which gives a balanced performance between compression efficiency and reconstructed quality with less drift. Our drift control mechanism maintains two frame buffers in the encoder and decoder; one that is based on the base layer and one that is based on the base plus enhancement layers. Drift control is achieved by switching between these two buffers for motion estimation and compensation. Our prediction is initially based on the base plus enhancement layers buffer, which inherently introduces drift in the system if a part of the enhancement layer is not available at the receiver. A measure of drift is computed based on the channel information and a threshold is set. When the measure exceeds the threshold, i.e., when drift becomes significant, we switch the prediction to be based on the base layer buffer, which is always available to the receiver. We also developed an adaptive scheme with additional computation overhead at the encoder to decide the switching instance. The performance of the threshold case that needs fewer computations is comparable with the adaptive scheme. Our coder offers high compression efficiency and sustained video quality for variable bit rate wireless channels. This proves that we need not completely eliminate drift and decrease compression efficiency to get better received video quality.     

TRASS: A transmission rate‐adapted streaming server in a wireless environment

The video streaming quality in a wireless communication network environment is largely affected by various network characteristics, such as a limited channel bandwidth and a variant transmission rate. The playback quality of User Equipments (UEs) may not be smooth when the service is delivered via a wireless environment. From the viewpoints of most video receivers, a smooth playback with a lower video quality may be more significant than a lagged or distorted playback with a higher video quality as the transmission rate degrades. Based on the above, we sketch an adaptation agent—Transmission‐Rate Adapted Streaming Server (TRASS), which is located between the original video server and UEs, to adaptively transform the streaming video based on the real transmission rate. In our proposed scheme, UEs would feedback their network access statuses to TRASS and then TRASS would deliver adaptive quality of video streams to UEs according to their feedbacks. The theoretical analysis and simulations using different video tracks encoded in MPEG‐4 and H.264/AVC formats show that TRASS can help wireless streaming users to get a smooth playback quality with a lower packet failure rate. With a low probability of receiving a worse quality of video, users' Quality of Experience can subsequently be raised. Copyright © 2010 John Wiley & Sons, Ltd.     

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.     

New video discarding policies for improving UDP performance over wired/wireless networks

Multimedia communication over wired and wireless networks becomes a compulsory need for many recent applications. To effectively react to the tremendous demand of video streaming over the Internet, videos are usually compressed by utilizing spatial and temporal redundancy. It is noteworthy to mention that compressing videos may degrade their quality if it is not investigated properly. In other words, as a consequence of exploiting redundancies, frame dependencies emanate, which make discarding frames, because of occupying the whole capacity of network elements, have severe implications on the video quality. Furthermore, transmitting videos over capacity‐limited links owing to error‐prone channels, power constraints and bandwidth variations will severely affect the video quality. Additionally, as the current coding schemes are characterized by being able to afford high compression efficiency, sensitivity to packet losses becomes untolerated. Therefore, insuring the perceived quality of the delivered videos to be always high in spite of aforementioned challenges is the primary focus of current researchers. In this paper, we propose efficient and novel video discarding policies that mainly aim to reduce the number of frames being lost through substitution of those frames that are very difficult or even impossible to decode at the receiver side. This is accomplished by controlling and maintaining the buffer occupancy of network elements. Our proposed policies are evaluated in terms of frameput, rate of non‐decodable frames, peak signal‐to‐noise ratio, structural similarity index and average buffer occupancy. Our proposed policies behave very well and achieve a remarkable enhancement over what is closely connected in the literature. Copyright © 2015 John Wiley & Sons, Ltd.     

Quality of experience (QoE) driven adaptation scheme for?voice/video over IP

Network quality of service (NQoS) of IP networks is unpredictable and impacts the quality of networked multimedia services. Adaptive voice and video schemes are therefore vital for the provision of voice over IP (VoIP) services for optimised quality of experience (QoE). Traditional adaptation schemes based on NQoS do not take perceived quality into consideration even though the user is the best judge of quality. Additionally, uncertainties inherent in NQoS parameter measurements make the design of adaptation schemes difficult and their performance suboptimal. This paper presents a QoE-driven adaptation scheme for voice and video over IP to solve the optimisation problem to provide optimal QoE for networked voice and video applications. The adaptive VoIP architecture was implemented and tested both in NS2 and in an Open IMS Core network to allow extensive simulation and test-bed evaluation. Results show that the scheme was optimally responsive to available network bandwidth and congestion for both voice and video and optimised delivered QoE for different network conditions, and is friendly to TCP traffic.