Path diversity for enhanced media streaming

Media streaming over best effort packet networks such as the Internet is quite challenging because of the dynamic and unpredictable available bandwidth, loss rate, and delay. Recently, streaming over multiple paths to provide path diversity has emerged as an approach to help overcome these problems. This article provides an overview of the benefits and use of path diversity for media streaming. The different approaches to media coding and streaming over multiple paths are examined, together with architectures for achieving path diversity between single or multiple senders and a single receiver. Important examples include using the distributed servers in a content delivery network to provide path diversity to a requesting client, using multiple 802.11 wireless access points to provide path diversity to a mobile client, and using relays to provide low-latency media communication. The design, analysis, and operation of media streaming systems that use path diversity are considered, with emphasis on the accurate performance models needed to select the best paths or best servers.     

Optimal multicast smoothing of streaming video over the Internet

A set of applications such as Internet video broadcasts, corporate telecasts, and distance learning require the simultaneous streaming of video to a large population of viewers across the Internet. The high bandwidth requirements and the multi-timescale burstiness of compressed video make it a challenging problem to provision network resources for streaming multimedia. For such applications to become affordable and ubiquitous, it is necessary to develop scalable techniques to efficiently stream video to a large number of disparate clients across a heterogeneous Internet. In this paper, we propose to multicast smoothed video over an application-level overlay network of proxies, and to differentially cache the video at the intermediate nodes (proxies) in the distribution tree, in order to reduce the network bandwidth requirements of video dissemination. We formulate the multicast smoothing problem as an optimization problem, and develop an algorithm for computing the set of transmission schedules for the tree that minimize the peak rate and rate variability, given buffer constraints at different nodes in the tree. We also develop an algorithm to compute the minimum buffer allocation in the entire tree, such that feasible transmission to all the clients is possible, when the tree has heterogeneous rate constraints. We show through trace-driven simulations that substantial benefits are possible from multicast smoothing and differential caching, and that these gains can be realized even with modest proxy caches.     

Ad Hoc上的新协议——流实时传输协议

在Ad Hoc网络中传输多媒体数据会受到网络拓扑变化和无线信道干扰的限制.多流实时传输协议MRTP(multi-flow real-time transport protocol)是一种可以分配和传输多个视频数据流的新协议.它结合多描述编码和多径传输,能提高无线对等网络上视频传输的效率和质量.     

Cross-layer architecture for scalable video transmission in wireless network

Multimedia applications such as video conference, digital video broadcasting (DVB), and streaming video and audio have been gaining popularity during last years and the trend has been to allocate these services more and more also on mobile users. The demand of quality of service (QoS) for multimedia raises huge challenges on the network design, not only concerning the physical bandwidth but also the protocol design and services. One of the goals for system design is to provide efficient solutions for adaptive multimedia transmission over different access networks in all-IP environment. The joint source and channel coding (JSCC/D) approach has already given promising results in optimizing multimedia transmission. However, in practice, arranging the required control mechanism and delivering the required side information through network and protocol stack have caused problems and quite often the impact of network has been neglected in studies. In this paper we propose efficient cross-layer communication methods and protocol architecture in order to transmit the control information and to optimize the multimedia transmission over wireless and wired IP networks. We also apply this architecture to the more specific case of streaming of scalable video streams. Scalable video coding has been an active research topic recently and it offers simple and flexible solutions for video transmission over heterogeneous networks to heterogeneous terminals. In addition it provides easy adaptation to varying transmission conditions. In this paper we illustrate how scalable video transmission can be improved with efficient use of the proposed cross-layer design, adaptation mechanisms and control information.     

Server-Aided Adaptive Live Video Streaming Over P2P Networks

Three factors, including churn of peers, high transmission delay, and high bandwidth heterogeneity, jointly bring forward great challenges to video streaming over P2P networks. In this paper, the multi-tree approach is leveraged to construct an overlay with resilience to churn and low transmission delay. For such a multi-tree structured overlay, a server-aided adaptive video streaming scheme is proposed to cope with the bandwidth heterogeneity. During streaming process, video data are collaboratively forwarded to the same receiver by multiple peers based on side information and network condition, as well as the distributed bitstream is dynamically switched among multiple available versions in a rate-distortion optimized way by the streaming server. Simulation results show that the proposed scheme achieves great gain in overall perceived quality over simple heuristic schemes.     

P2P流媒体中的多发送端选择算法

 在多对单P2P流模式下,如何选择多个发送端,并为其最优地分配发送速率和数据是一个难题.为此,提出了一种新的多发送端选择和最优速率及数据分配算法(MSRDA).首先把待解决的问题模型化为线性最优化问题,然后给出了求解最优化问题的算法.不同于已有的算法只按链路的丢包率来选择发送端,MSRDA根据到各发送端链路的丢包率与链路的可用带宽比升序排序来选择多发送端,并最优地分配发送速率和数据.理论分析和仿真实验结果表明:在不同的网络条件下MSRDA能有效地减少聚合带宽的丢包率,并具有自适应性,优于现有同类算法.     

Adaptive streaming of multi-view video over P2P networks

In this paper, we propose a novel solution for the adaptive streaming of 3D representations in the form of multi-view video by utilizing P2P overlay networks to assist the media delivery and minimize the bandwidth requirement at the server side. Adaptation to diverse network conditions is performed regarding the features of human perception to maximize the perceived 3D. We have performed subjective tests to characterize these features and determined the best adaptation method to achieve the highest possible perceived quality. Moreover, we provide a novel method for mapping from scalable video elementary stream to torrent-like data chunks for adaptive video streaming and provide an optimized windowing mechanism that ensures timely delivery of the content over yanl?? gibi. The paper also describes techniques generating scalable video chunks and methods for determining system parameters such as chunk size and window length.     

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.     

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.     

Frame distortion estimation for unequal error protection methods in scalable video coding (SVC)

In many multimedia applications, coded video is transmitted over error prone heterogeneous networks. Because of the predictive mechanism used in video coding, transmission error would propagate temporally and spatially and would result in significant quality losses. In order to address this problem, different error resilience methods have been proposed. One of the techniques, which is commonly used in video streaming, is unequal error protection (UEP) of scalable video coding (SVC). In this technique, different independent layers of an SVC stream are protected differently and based on their importance by using forward error correction (FEC) codes. Accurately analyzing the importance or utility of each video part is a critical component and would lead to a better protection and higher quality of the received video. Calculation of the utility is usually based on multiple decoding of sub-bitstreams and is highly computationally complex. In this work, we propose an accurate low complexity utility estimation technique that can be used in different applications. This technique estimates the utility of each network abstraction layer (NAL) by considering the error propagation to future frames. We utilize this method in an UEP framework with the scalable extension of H.264/AVC codec and it achieves almost the same performance as highly complex estimation techniques (an average loss of 0.05 dB). Furthermore, we propose a low delay version of this technique that can be used in delay constrained application. The estimation accuracy and performance of our proposed technique are studied extensively.     

Video streaming over P2P networks: Challenges and opportunities

A robust real-time video communication service over the Internet in a distributed manner is an important challenge, as it influences not only the current Internet structure but also the future Internet evolution. In this context, Peer-to-Peer (P2P) networks are playing an imperative position for providing efficient video transmission over the Internet. Recently, several P2P video transmission systems have been proposed for live video streaming services or video-on-demand services over the Internet. In this paper, we describe and discuss existing video streaming systems over P2P. Efficient (delay tolerant and intolerant) data sharing mechanisms in P2P and current video coding trends are elaborated in detail. Moreover, video streaming solutions (live and on-demand) over P2P from the perspective of tree-based and mesh-based systems are explained. Finally, the conclusion is drawn with key challenges and open issues related to video streaming over P2P.     

Efficient slice‐aware H.264/AVC video transmission over time‐driven priority networks

This paper proposes a multimedia streaming architecture that combines network and video technologies to handle video traffic over multi‐hop access networks. In this context, resource overprovision typical of current quality of service approaches will become a limiting factor because of the increasing spread of bandwidth‐intensive multimedia applications. The proposed scheme adopts a time‐driven priority scheduling at network nodes and exploits slice classification at the video encoder to differentiate packets. The service guarantees offered by time‐driven priority, together with the packet classification, significantly reduce congestion and increase the video quality at the receiver with respect to the traditional differentiated services (DiffServ) approach, also achieving high resource utilization. This is an important result, as current DiffServ‐based architectures are far from obtaining such resource utilization in conjunction with reasonable delays and jitters. Copyright © 2013 John Wiley & Sons, Ltd.     

P2P环境下支持版权交易的DRM模型

对等网络因其具有分布式、自组织、带宽利用率高等特点,在分布式文件分发、多媒体传输、网络流媒体服务方面有着广阔的应用前景,成为研究领域的一个热点。但是由于对等网络分布式的结构特点,必须引入版权保护机制对多媒体文件版权进行保护。文章提出了一种P2P环境下支持版权交易的DRM模型,为解决对等网络下的版权交易问题提出了一个可行的方案。     

A buffer‐aware fleet‐based cooperative H.264/SVC streaming over vehicular networks

Because of of the characteristics of high mobility, time varying and dynamic topology, how to provide multimedia streaming service for connected vehicles becomes one emerging and popular technical research. The motivation of this paper is to utilize cooperation among neighboring vehicles for video streaming's quality improvement over vehicular networks. In the proposed cooperative streaming scenario, a connected vehicle requests a video stream from the Internet by using its 3G/3.5G interface, which may not have enough bandwidth to receive good quality of video. Thus, the vehicle is suggested to ask neighboring members belonging to the same fleet to download the requested video data by using their 3G/3.5G interfaces. Then, neighboring members should forward video data to the requested vehicle by using another wireless technique, for example, dedicated short range communication (DSRC). Regarding the differentiation between the two access networks, that is, 3G/3.5G network and DSRC network, a buffer‐aware scheduling mechanism based on layered streaming is designed in this paper to adapt to the networking situation of the vehicular networks. Two selection algorithms are proposed to select neighboring vehicles from the fleet. According to our simulation results, the 3G/3.5G‐based selection algorithm is suitable to improve video quality for vehicles at low speeds. On the other hand, the DSRC‐based selection algorithm can get better performance when vehicles move at high speeds or too many data are transmitted among vehicles.Copyright © 2013 John Wiley & Sons, Ltd.     

Streaming Multimedia for the Internet

Multimedia streaming enables Web sites to be enhanced with real-time audio and video content. Through the appropriate use of video compression technologies, audio and video content can be delivered directly into the Web browser environment to create a rich mix of text, graphics and multimedia. The audio quality and video window size available to viewers is primarily determined by network access speeds. For the Internet, audio and video can be successfully streamed at rates between 20 kbit/s and 37 kbit/s over a PSTN modem link. In the corporate intranet environment, where higher end-to-end network speeds exist, audio and video can easily be streamed at rates between 56 kbit/s and 450 kbit/s. This paper illustrates how commodity video compression techniques facilitate the introduction of real-time multimedia streaming over IP networks. It describes the technology components within a basic video streaming system and provides an overview of two applications that utilise this technology, namely Intranet TV and BT Results Webcast. The underlying network protocols required to support real-time Internet applications are also discussed and network measurement statistics obtained from Intranet TV and BT Results Webcast are also presented.     

A Novel End-to-End Architecture for H.264 Video Streaming over the Internet

In this paper we propose a novel end-to-end architecture for H.264 unicast video streaming over the Internet. The proposed video streaming architecture is based primarily on a new transport layer protocol, the stream control transmission protocol (SCTP). We show that the network-friendly specification of H.264, and the novel technical characteristics of SCTP, when coupled together are able to provide a highly adaptive and flexible system for unicast video streaming. More specifically, we develop algorithms that handle at the transport layer the following functions concerning video packet transmission: retransmission policy, packet prioritization, implicit receiver feedback. We combine the above algorithms with an R-D optimization strategy at the encoder, that provides more options when adapting to bandwidth variations. This combined optimization strategy leads to more options concerning the streaming parameters. Finally, with simulation results we prove that our system is capable of maintaining good perceptual quality and TCP-friendliness under various loss conditions. An early version of this paper appeared in CCNC 2004.