Adaptive systems for improved media streaming experience

Supporting streaming media applications over current packet network infrastructures represents a challenging task in many regards. For one, the lack of quality of service (QoS) guarantees in existing networks such as the Internet means that time-constrained media packets will face dynamic variations in bandwidth, loss rate, and delay as they traverse the network from the sender to the receiver. The variable rate of media traffic represents yet another difficulty when transmission constraints need to be met. Finally, the heterogeneity of client devices and access bandwidth coupled with custom user preferences exacerbate the problem of smooth and quality-optimized media playback even further. In this article we provide an overview of the various techniques for media and streaming strategy adaptation, which can be employed to deal with the difficulties imposed by such dynamic environments. These techniques depend on the characteristics of the media application, in particular on the network streaming infrastructure and the timing constraints imposed on the media packets' delivery. We survey adaptation techniques that act on the encoding of the multimedia information, on the scheduling of the media packets, or that try to combat transmission errors. We also briefly overview some media-friendly networking solutions, which contribute to increased QoS by incorporating some level of intelligence in intermediate network nodes. Finally, we describe a few open challenges in media streaming, emphasizing strategies based on promising cross-layer approaches where adaptation strategies are applied in a coordinated manner, across different layers of the network protocol stack     

基于无线网络的实时视频传输系统的设计

通过对流媒体技术和网络协议的深入研究,设计和实现了一种基于RTP(实时传输协议)和FEC(前向纠错)的无线视频实时传输系统.该系统采用C/S(Client/Server)架构,服务器端基于RTP协议对数据包进行封装,并加入FEC纠错机制;客户端根据RTP数据包的头部信息,重建包序列,定位丢失的数据包,并利用FEC算法对丢失数据进行恢复.实际测试证明该系统能够在无线网络中有效地保证接收端数据的实时性和完整性.     

RaDiO Edge: Rate-Distortion Optimized Proxy-Driven Streaming From the Network Edge

This paper addresses the problem of streaming packetized media over a lossy packet network through an intermediate proxy server to a client, in a rate-distortion optimized way. The proxy, located at the junction of the backbone network and the last hop to the client, coordinates the communication between the media server and the client using hybrid receiver/sender-driven streaming in a rate-distortion optimization framework. The framework enables the proxy to determine at every instant which packets, if any, it should either request from the media server or (re)transmit directly to the client, in order to meet constraints on the average transmission rates on the backbone and the last hop while minimizing the average end-to-end distortion. Performance gains are observed over rate-distortion optimized sender-driven systems for streaming packetized video content. The improvement in performance depends on the quality of the network path both in the backbone network and along the last hop     

An efficient network coded ARQ for multisource multidestination relay networks over mixed flat fading channels

This paper proposes a new reliable automatic repeat request (ARQ) transmission protocol for wireless multisource multidestination relay networks over mixed fading channels. Conventional application of ARQ protocols to retransmit lost or erroneous packets in relay networks can cause considerable delay latency with a significant increase in the number of retransmissions when networks consist of multiple sources and multiple destinations. To address this issue, a new ARQ protocol based on network coding (NC) is proposed where the relay detects packets from different transmission sources, then uses NC to combine and forward lost packets to their destinations. An efficient means for the retransmission of all lost packets is proposed through two packet-combination algorithms for retransmissions at the relay and sources. The paper derives mathematical formulation of transmission bandwidth for this new NC-based ARQ protocol and compares analytical and simulation results with some other ARQ protocols over both mixed Rayleigh and Rician flat fading channel. The mixed fading model permits investigation of two typical fading scenarios where the relay is located in the neighbourhood of either the sources or the destinations. The transmission bandwidth results show that the proposed NC-based ARQ protocol demonstrates superior performance over other existing ARQ schemes.     

Waiting time and queue length distributions for go-back-N andselective-repeat ARQ protocols

The paper provides a performance analysis of automatic-repeat-request (ARQ) protocols in connection-oriented transmission. Each message arriving at a transmitter is divided into several packets, which are continuously transmitted to a receiver according to go-back-N ARQ or selective-repeat ARQ protocols. Because of connection-oriented transmission, messages are served on a first come first served basis, i.e., transmission of a message is commenced after all packets in the previous message are successfully transmitted. For the two ARQ protocols, the authors derive the probability generating functions of message waiting time and queue length at an arbitrary instant, from which the average performance measures are explicitly obtained     

Multi-path content delivery: Efficiency analysis and optimization algorithms

The paper studies the benefits of multi-path content delivery from a rate-distortion efficiency perspective. We develop an optimization framework for computing transmission schedules for streaming media packets over multiple network paths that maximize the end-to-end video quality, for the given bandwidth resources. We comprehensively address the two prospective scenarios of content delivery with packet path diversity. In the context of sender-driven systems, our framework enables the sender to compute at every transmission instance the mapping of packets to network paths that meets a rate constraint while minimizing the end-to-end distortion. In receiver-driven multi-path streaming, our framework enables the client to dynamically decide which packets, if any, to request for transmission and from which media servers, such that the end-to-end distortion is minimized for a given transmission rate constraint. Via simulation experiments, we carefully examine the performance of the scheduling framework in both multi-path delivery scenarios. We demonstrate that the optimization framework closely approaches the performance of an ideal streaming system working at channel capacity with an infinite play-out delay. We also show that the optimization leads to substantial gains in rate-distortion performance over a conventional content-agnostic scheduler. Through the concept of error-cost performance for streaming a single packet, we provide another useful insight into the operation of the optimization framework and the conventional scheduling system.     

Erasure coding for reliable adaptive retransmission in wireless broadcast/multicast systems

In this paper, we present new adaptive automatic repeat request (ARQ) schemes for wireless broadcast/multicast combining erasure coding (EC) and packet retransmission. Traditional approaches rely on retransmitting the lost packets in a point-to-point or point-to-multipoint mode. The main idea behind the presented protocols is to retransmit adaptive combinations of the lost packets using EC, which can help several receivers to recover the lost information with fewer retransmission attempts. We propose two versions of EC-based ARQ protocols, and investigate theoretically the corresponding transmission bandwidths in different contexts. We show through simulation results the efficiency of the proposed protocols with respect to conventional ARQ strategies and new published ARQ works for broadcast/multicast. Finally, a new sliding window NACK feedback policy is presented for the case of a high number of receivers to avoid the feedback implosion problem.     

Analytical framework for simultaneous MAC packet transmission (SMPT) in a multicode CDMA wireless system

Stabilizing the throughput over wireless links is one of the key challenges in providing high-quality wireless multimedia services. Wireless links are typically stabilized by a combination of link-layer automatic repeat request (ARQ) mechanisms in conjunction with forward error correction and other physical layer techniques. In this paper, we focus on the ARQ component and study a novel class of ARQ mechanisms, referred to as simultaneous MAC packet transmission (SMPT). In contrast to the conventional ARQ mechanisms that transmit one packet at a time over the wireless air interface, SMPT exploits the parallel code channels provided by multicode code-division multiple access. SMPT stabilizes the wireless link by transmitting multiple packets in parallel in response to packet drops due to wireless link errors. While these parallel packet transmissions stabilize the link layer throughput, they also increase the interference level in a given cell of a cellular network or cluster of an ad hoc network. This increased interference reduces the number of traffic flows that can be simultaneously supported in a cell/cluster. We develop an analytical framework for the class of SMPT mechanisms and analyze the link-layer buffer occupancy and the code usage in a wireless system running some form of SMPT. Our analysis quantifies the tradeoff between increased link-layer quality of service and reduced number of supported flows in SMPT with good accuracy, as verified by simulations. In a typical scenario, SMPT reduces the probability of link-layer buffer overflow by over two orders of magnitude (thus enabling high-quality multimedia services, such as real-time video streaming) while supporting roughly 20% fewer flows than conventional ARQ. Our analytical framework provides a basis for resource management in wireless systems running some form of SMPT and optimizing SMPT mechanisms.     

基于细粒度调整的同步播放控制算法

针对网络时延和时延抖动引起缓存波动而造成的流媒体播放不连续,基于接收端播放速率调整,提出一种播放速率自适应调整的播放缓存控制算法。算法依据缓存区的占用水平,通过构造控制序列,动态、细粒度地调整媒体的播放速率。在校园网环境下的模拟结果表明,该算法能够有效地减少缓存上溢或下溢引起的播放跳跃或停顿,从而实现流媒体的平滑播放。     

Weighted partial network coding and its applications in wireless mesh networks

Network coding (NC) has showed to be beneficial to improve transmission performance in wireless mesh networks. Random linear coding is usually applied as the default coding schema. However, random linear coding causes significant decoding delay and jitter at receiver. Further, current NC does not support weight assignment to original packets, which is however indispensable for popular applications such as quality of service control and multipath media streaming in wireless mesh networks. Partial network coding (PNC) can largely reduce decoding delay and receiving fluctuation while keeping the benefit of NC. However, PNC does not support weight‐based data replacement and weight assignment to original packets. In this work, we propose weighted partial network coding (WPNC), which is a generalized coding schema of PNC. WPNC inherits all merits of PNC and part of NC. With WPNC, both decoding delay and receiving fluctuation will be reduced as observed in PNC. Also, WPNC is quite suitable for those applications that require weight assignment to original packets. After providing the whole framework of WPNC and thorough theoretical analysis to its performance, we have demonstrated how WPNC can be integrated with quality of service control and multipath routing supported media streaming in wireless mesh networks. Performance of WPNC is inter‐validated by both theoretical analysis and numeric evaluations. Copyright © 2011 John; Wiley & Sons, Ltd.     

基于5G通信技术的低延时网络直播解决方案

在媒体数字化转型进程中,互联网与通信技术的创新应用是融媒体生态体系的重要推动力.本文结合媒体直播的行业背景与所面临的实际挑战,介绍了由河南广播电视台自主研发建设的大象图传超高清信号传输制播平台.平台基于5G通信技术和超高清视频编码技术,实现了广播级超高清视频直播与互动,为媒体直播产业的发展提供了一种新的解决方案.     

Optimal Cross-Layer Scheduling of Transmissions Over a Fading Multiaccess Channel

We consider the problem of several users transmitting packets to a base station, and study an optimal scheduling formulation involving three communication layers, namely, the medium access control, link, and physical layers. We assume Markov models for the packet arrival processes and the channel gain processes. Perfect channel state information is assumed to be available at the transmitter and the receiver. The transmissions are subject to a long-run average transmitter power constraint. The control problem is to assign power and rate dynamically as a function of the fading and the queue lengths so as to minimize a weighted sum of long run average packet transmission delays.     

Scalable on-demand media streaming with packet loss recovery

Previous scalable on-demand streaming protocols do not allow clients to recover from packet loss. This paper develops new protocols that: (1) have a tunably short latency for the client to begin playing the media; (2) allow heterogeneous clients to recover lost packets without jitter as long as each client's cumulative loss rate is within a tunable threshold; and (3) assume a tunable upper bound on the transmission rate to each client that can be as small as a fraction (e.g., 25%) greater than the media play rate. Models are developed to compute the minimum required server bandwidth for a given loss rate and playback latency. The results of the models are used to develop the new protocols and assess their performance. The new protocols, Reliable Periodic Broadcast and Reliable Bandwidth Skimming, are simple to implement and achieve nearly the best possible scalability and efficiency for a given set of client characteristics and desirable/feasible media quality. Furthermore, the results show that the new reliable protocols that transmit to each client at only twice the media play rate have similar performance to previous protocols that require clients to receive at many times the play rate.     

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.     

Adaptation strategies for MGS scalable video streaming

An adaptive streaming framework consists of a video codec that can produce video encoded at a variety of rates, a transport protocol that supports an effective rate/congestion control mechanism, and an adaptation strategy in order to match the video source rate to the available network throughput. The main parameters of the adaptation strategy are encoder configuration, video extraction method, determination of video extraction rate, send rate control, retransmission of lost packets, decoder buffer status, and packetization method. This paper proposes optimal adaptation strategies, in terms of received video quality and used network resources, at the codec and network levels using a medium grain scalable (MGS) video codec and two transport protocols with built-in congestion control, TCP and DCCP. Key recommendations are presented to obtain the best results in adaptive video streaming using TCP or DCCP based on extensive experimental results over the Internet.     

An altruistic cross‐layer recovering mechanism for ad hoc wireless networks

Video streaming services have restrictive delay and bandwidth constraints. Ad hoc networks represent a hostile environment for this kind of real‐time data transmission. Emerging mesh networks, where a backbone provides more topological stability, do not even assure a high quality of experience. In such scenario, mobility of terminal nodes causes link breakages until a new route is calculated. In the meanwhile, lost packets cause annoying video interruptions to the receiver. This paper proposes a new mechanism of recovering lost packets by means of caching overheard packets in neighbor nodes and retransmit them to destination. Moreover, an optimization is shown, which involves a video‐aware cache in order to recover full frames and prioritize more significant frames. Results show the improvement in reception, increasing the throughput as well as video quality, whereas larger video interruptions are considerably reduced. Copyright © 2014 John Wiley & Sons, Ltd.     

Resequencing delay and buffer occupancy under the selective-repeatARQ

Consider a communication network that regulates retransmissions of erroneous packets by a selective-repeat (SR) automatic repeat request (ARQ) protocol. Packets are assigned consecutive integers, and the transmitter continuously transmits them in order until a negative acknowledgement or a time-out is observed. The receiver, upon receipt of a packet, checks for errors and returns positive/negative acknowledgement (ACK/NACK) accordingly. Only packets for which either NACK or time-out have been observed are retransmitted. Under SR ARQ, the receiver accepts packets that are out of order and must store them temporarily if it has to deliver them in sequence. The resequencing buffer requirements and the resulting packet delay constitute major factors in overall system considerations. The authors derive the distributions of the buffer occupancy and the resequencing delay at the receiver under a heavy traffic situation. This enables the network designer to determine how much buffer capacity at the receiver guarantees certain specified performance measures     

Analysis of packet loss in transport layer over ATM networks

In asynchronous transfer mode (ATM) networks, when cells are lost due to congestion, packets containing the lost cells should be retransmitted in the transport layer, which manages the end-to-end communication. The probability that a packet contains at least one lost cell depends on the packet length. It is thus very likely that the performance of the end-to-end communication is influenced by the packet length. In this paper, we analyse packet loss probability and the achievable maximum throughput when a block of data is divided into packets of fixed size and the lost packets are retransmitted based on the selective repeat automatic repeat request (ARQ). Through this analysis, we examine the effect of packet length and peak cell transmission rate on the performance measures mentioned above. © 1997 John Wiley & Sons, Ltd.     

基于机会式网络编码的低时延广播传输算法

为了提高无线网络中数据包广播传输的效率,本文提出了一种基于机会式网络编码的广播传输算法.该算法在发送端按一定顺序选择不同终端的丢包,并采用异或运算编码重传包,在终端采用从重传包中解码数据包的方法恢复丢包.该算法优先恢复时间重要性较高的丢包,并使多个终端同时从单个重传包恢复其丢包,因此有效地提高了广播传输效率并降低了传输...