Impact of device unavailability on the reliability of multicast transport in IEEE 802.11 networks

Multicast transport is an efficient solution to deliver the same content simultaneously to many receivers. This transport mode is mainly used these days to deliver real-time video streams. However, multicast transmissions support over IEEE 802.11 networks does not provide any feedback policies, which implies a definite loss of missing packets. This impacts the reliability of the multicast transport and the application employing it. An alternative to improve the reliability of multicast streaming over 802.11 networks is to prevent packet losses. In this perspective, it is necessary to identify the loss causes and to perform the required prevention actions. It is well known that collisions and path loss are two fundamental sources of transmission failures. Their impact can be eliminated by means of collision prevention and data rate adaptation. However, several works show that the loss rate of multicast packets may be considerable even in collisions-free environments and using an appropriate transmission rate. Particularly they show that losses may have a bursty nature which does not correspond to the bit error rate model of the PHY layer as defined by the chipset manufacturers. Therefore, in this paper, we carry out a thorough investigation of the loss causes in wireless networks. We show that device unavailability may be the principal cause of the significant packet losses that occur and their bursty nature. Particularly, our results show that the CPU overload may incur a loss rate of 100%, and that the delivery ratio may be limited to 35% when the device is in the power save mode.     

基于区分服务的流媒体传输模型

流媒体在网络上的应用经常会受到数据包丢失或错误以及网络带宽资源不足的干扰,使得接收方播放质量受到严重影响。本文建立了一种适合流媒体传输的区分服务模型,该模型能够使高优先级数据流(实时流媒体数据流)占用更多的带宽。仿真结果表明:该策略能使流媒体数据流获得较高的吞吐量和较低的丢包率,有效提高流媒体传输的可靠性和实时性。     

基于区分服务的流媒体传输模型

流媒体在网络上的应用经常会受到数据包丢失或错误以及网络带宽资源不足的干扰,使得接收方播放质量受到严重影响。本文建立了一种适合流媒体传输的区分服务模型,该模型能够使高优先级数据流（实时流媒体数据流）占用更多的带宽。仿真结果表明：该策略能使流媒体数据流获得较高的吞吐量和较低的丢包率,有效提高流媒体传输的可靠性和实时性。     

Rate-distortion optimized distributed packet scheduling of multiple video streams over shared communication resources

We consider the problem of distributed packet selection and scheduling for multiple video streams sharing a communication channel. An optimization framework is proposed, which enables the multiple senders to coordinate their packet transmission schedules, such that the average quality over all video clients is maximized. The framework relies on rate-distortion information that is used to characterize a video packet. This information consists of two quantities: the size of the packet in bits, and its importance for the reconstruction quality of the corresponding stream. A distributed streaming strategy then allows for trading off rate and distortion, not only within a single video stream, but also across different streams. Each of the senders allocates to its own video packets a share of the available bandwidth on the channel in proportion to their importance. We evaluate the performance of the distributed packet scheduling algorithm for two canonical problems in streaming media, namely adaptation to available bandwidth and adaptation to packet loss through prioritized packet retransmissions. Simulation results demonstrate that, for the difficult case of scheduling nonscalably encoded video streams, our framework is very efficient in terms of video quality, both over all streams jointly and also over the individual videos. Compared to a conventional streaming system that does not consider the relative importance of the video packets, the gains in performance range up to 6 dB for the scenario of bandwidth adaptation, and even up to 10 dB for the scenario of random packet loss adaptation.     

Distributed Collaboration for Enhanced Sender-Driven Video Streaming

We propose a sender-driven system for adaptive streaming from multiple servers to a single receiver over separate network paths. The servers employ information in receiver feedbacks to estimate the available bandwidth on the paths and then compute appropriate transmission schedules for streaming media packets to the receiver based on the bandwidth estimates. An optimization framework is proposed that enables the senders to compute their transmission schedules in a distributed way, and yet to dynamically coordinate them over time such that the resulting video quality at the receiver is maximized. To reduce the computational complexity of the optimization framework an alternative technique based on packet classification is proposed. The substantial reduction in online complexity due to the resulting packet partitioning makes the technique suitable for practical implementations of adaptive and efficient distributed streaming systems. Simulations with Internet network traces demonstrate that the proposed solution adapts effectively to bandwidth variations and packet loss. They show that the proposed streaming framework provides superior performance over a conventional distortion-agnostic scheme that performs proportional packet scheduling on the network paths according to their respective bandwidth values.     

Concurrent multipath transmission combining forward error correction and path interleaving for video streaming

Video streaming is a popular application on next generation networks (NGNs). However, video streaming over NGNs has many challenges due to the high bit error rates of these networks. Forward error correction (FEC) is often applied to improve the quality of video streaming. However, continuous lost packets decrease the recovery performance of FEC protection in NGNs. To disperse continuous lost packets to different FEC blocks, we propose a concurrent multipath transmission that combines FEC with path interleaving. Our proposed control scheme adaptively adjusts the FEC block length and concurrently sends data interleaved over multiple paths. Experimental results with our approach show improved packet loss and signal to noise ratio performance.     

一种适用于无线网络的流媒体传输机制

为保证无线网络中多媒体数据的传输质量,提出了一种适用于无线网络的流媒体传输机制（WMTCC）。该机制通过发送探测报文区分网络拥塞丢包和链路误码随机丢包,准确判断网络的拥塞状况,实施发送速率调节,保证了流媒体服务质量(QoS)。由于准确区分出无线链路误码丢包,该机制在链路误码率较高时能维持较高的网络吞吐量。仿真实验结果显示在高误码率无线网络中,该机制可以获得更高的吞吐量和更大的拥塞窗口,并且发送速率的变化更加平滑。     

Multi-path continuous media streaming: what are the benefits?

Quality of service (QoS) in delivery of continuous media (CM) over the Internet is still relatively poor and inconsistent. Although many such applications can tolerate some degree of missing information, significant losses degrade an application’s QoS. In this paper, we investigate the potential benefits of mitigating this problem through the exploitation of multiple paths existing in the network between a set of senders and a receiver of CM. Our focus in this work is on providing a fundamental understanding of the benefits of using multiple paths to deliver CM over best-effort wide-area networks. Specifically, we consider pre-recorded CM applications and use the following metrics in evaluating the performance of multi-path streaming as compared to single-path streaming: (a) data loss rate, (b) conditional error burst length distribution, and (c) lag1-autocorrelation. The results of this work can be used in guiding the design of multi-path CM systems streaming data over best-effort wide-area networks.     

基于RTMP的流媒体服务器的研究

多媒体数据要在网络上传输,必须先对多媒体数据进行流化处理。要想流化多媒体数据,就需要流媒体服务器。所以流媒体服务器对多媒体数据的传输有着至关重要作用。其中流化处理就是对多媒体数据进行封装,把音视频数据打包成能进行流传输的数据包。而且提高流媒体服务器的性能一个有效途径是提高缓存的利用率,使系统为更多的媒体流服务。文章以RTMP为基础,首先分析了该传输协议,并提出了一种改进的缓存策略,使流媒体服务器性能有了提高。     

流媒体自适应传输控制的研究

流媒体是指多媒体数据流在网络上一边传输一边播放的一种多媒体通信服务.提供尽力而为服务的Internet不能为流媒体保证网络带宽、传输延迟、分组丢失以及分组错误等,而自适应传输控制机制能够提高流媒体服务的传输服务质量和传输服务的公平性.本文探讨流媒体自适应传输控制技术所涉及的各个方面,包括拥塞控制、质量自适应和错误控制.     

Cross-Layer prioritized H.264 video packetization and error protection over noisy channels

Video transmission over wireless channels is affected by channel-induced packet losses. Distortion due to channel errors can be alleviated by applying forward error correction. Aggregating H.264/AVC slices to form video packets with sizes adapted to their importance can also improve transmission reliability. Larger packets are more likely to be in error but smaller packets require more overhead. We present a cross-layer dynamic programming (DP) approach to minimize the expected received video distortion by jointly addressing the priority-adaptive packet formation at the application layer and rate compatible punctured convolutional (RCPC) code rate allocation at the physical layer for prioritized slices of each group of pictures (GOP). Some low priority slices are also discarded to improve protection to more important slices and meet the channel bit-rate limitations. We propose two schemes. Our first scheme carries out joint optimization for all slices of a GOP at a time. The second scheme extends our cross-layer DP-based approach to slices of each frame by predicting the expected channel bit budget per frame for live streaming. The prediction uses a generalized linear model developed over the cumulative mean squared error per frame, channel SNR, and normalized compressed frame bit budget. The parameters are determined over a video dataset that spans high, medium and low motion complexity. The predicted frame bit budget is used to derive the packet sizes and corresponding RCPC code rates for live transmission using our DP-based approach. Simulation results show that both proposed schemes significantly improve the received video quality over contemporary error protection schemes.     

面向TCP的流媒体传输编码码率自适应算法

由于网络的时变性和异构性,以及在拥塞情况下的高丢包率,利用TCP传输流媒体数据是Internet流媒体分发系统提高流媒体分发质量的首选方案。由于TCP具有超时或错误重传机制,在网络拥塞情况下,难以保证高码率流媒体数据传输的实时性,因此提出一种面向TCP流媒体传输的编码码率自适应算法(TCP_RA)。该算法根据流媒体发送应用层缓冲区读写指针差值调整流媒体发送端的编码码率适应网络带宽的变化。仿真实验对比分析了该算法与基于UDP之上TFRC协议的流媒体传输码率自适应算法在流媒体传输质量上的差别。结果表明,该算法在网络环境较差的情况下有效地提高了流媒体传输质量。并且该算法容易实现,值得推广。     

一种MPEG-4 FGS流媒体自适应传输系统

针对MPEG-4FGS流媒体提出一种质最自适应传输系统,采用3种质量平滑机制,即GOP质量平滑、帧质量平滑和FEC差错控制,保证在丢包情况下的连续GOP以及每个GOP内连续帧的质量稳定。模拟结果表明,自适应传输系统能和Internet丢包环境下平滑连续GOP的质量和GOP内连续帧的质量。     

Failure-Aware, Open-Loop, Adaptive Video Streaming With Packet-Level Optimized Redundancy

A plethora of coding and streaming mechanisms have been proposed for real-time multimedia transmission over the Internet. However, most proposed mechanisms rely only on global (e.g. based on end-to-end measurements), delayed (at least by the round-trip-time), or statistical (often based on simplistic network models) information available about the network state. Based on recently-proposed state-of-the-art open-loop video coding schemes, we propose a new integrated streaming and routing framework for robust and efficient video transmission over networks exhibiting path failures. Our approach explicitly takes into account the network dynamics, path diversity, and the modeled video distortion at the receiver side to optimize the packet redundancy and scheduling. In the derived framework, multimedia streams can be adapted dynamically at the video server based on instantaneous routing-layer information or failure-modeling statistics. The performance of our integrated application and network-layer method is simulated against equivalent approaches that are not optimized based on routing-layer feedback and distortion modeling, and the obtained gains in video quality are quantified     

Fine-Granularity Transmission Distortion Modeling for Video Packet Scheduling Over Mesh Networks

Packet scheduling is a critical component in multi-session video streaming over mesh networks. Different video packets have different levels of contribution to the overall video presentation quality at the receiver side. In this work, we develop a fine-granularity transmission distortion model for the encoder to predict the quality degradation of decoded videos caused by lost video packets. Based on this packet-level transmission distortion model, we propose a content-and-deadline-aware scheduling (CDAS) scheme for multi-session video streaming over multi-hop mesh networks, where content priority, queuing delays, and dynamic network transmission conditions are jointly considered for each video packet. Our extensive experimental results demonstrate that the proposed transmission distortion model and the CDAS scheme significantly improve the performance of multi-session video streaming over mesh networks.     

基于视频发送端速率控制的QoS技术研究

基于尽力而为的网络模式不能提供QoS保证,网络拥塞和分组丢失不可避免。在端到端视频单播结构下,论文提出了一个发送端速率控制框架SRCF,在此框架下首先利用RTCP报文中的字段提出了一种网络参数测量方法,然后设计了一个自适应速率算法SRCA,SRCA利用已得到的网络传输延迟和分组丢失率参数作为初始参数,来调整编码速率,达到充分利用带宽的目的,避免了视频质量由于调整参数带来的剧烈抖动。仿真结果表明,该算法在网络出现一定拥塞的条件下,能跟踪带宽的变化,网络和媒体QoS能保证视频质量较好。     

An efficient delay-constrained ARQ scheme for MMT packet-based real-time video streaming over IP networks

The MPEG has recently Querydeveloped a new standard, MPEG media transport (MMT), for the next-generation hybrid media delivery service over IP networks considering the emerging convergence of digital broadcast and broadband services. On account of the heterogeneous characteristics of broadcast and broadband networks, MMT provides an efficient delivery timing model to enable inter-network synchronization, measure various kinds of transmission delays and jitters caused by the transmission delay, and re-adjust the timing relationship between the MMT packets to ensure synchronized playback. By exploiting the delivery timing model, it is possible to accurately estimate the round-trip time (RTT) experienced during MMT packet transmission. Based on the measured RTT, we propose an efficient delay-constrained automatic repeat request (ARQ) scheme, which is applicable to MMT packet-based real-time video streaming service over IP networks. In the proposed ARQ scheme, the receiver buffer fullness at the time of packet loss detection is used to compute the arrival deadline, which is the maximum allowed time for completing the requesting and retransmitting of the lost MMT packet. Simulation results demonstrate that the proposed delay-constrained ARQ scheme can not only provide reliable error recovery, but it also achieves significant bandwidth savings by reducing the number of wastefully retransmitted packets that arrive at the receiver side and exceed the allowed arrival deadline.     

Generic forward error correction of short frames for IP streaming applications

If the frame size of a multimedia encoder is small, Internet Protocol (IP) streaming applications need to pack many encoded media frames in each Real-time Transport Protocol (RTP) packet to avoid unnecessary header overhead. The generic forward error correction (FEC) mechanisms proposed in the literature for RTP transmission do not perform optimally in terms of stability when the RTP payload consists of several individual data elements of equal priority. In this paper, we present a novel approach for generating FEC packets optimized for applications packing multiple individually decodable media frames in each RTP payload. In the proposed method, a set of frames and its corresponding FEC data are spread among multiple packets so that the experienced frame loss rate does not vary greatly under different packet loss patterns. We verify the performance improvement gained against traditional generic FEC by analyzing and comparing the variance of the residual frame loss rate in the proposed packetization scheme and in the baseline generic FEC.     

基于实时传输协议的丢包实时修复

在诸如IP电话和远程会议系统等实时应用中,使用实时传输协议RTP(real-timetransportprotocol)在因特网上传输的数据包不可避免地会丢失,极大地影响了传输服务质量.在RTP上增加丢包修复功能可以解决这个问题.介绍了在RTP上采用FEC(forwarderrorcorrection)修复丢包的方法,基于这个方法设计并实现了一个支持丢包修复功能的RTP库,说明了丢包修复功能的测试方法和结果.     

A new fast friendly window-based congestion control for real-time streaming media transmission

1 Introduction In the current Internet, not all applications use TCP and they do not follow the same concept of fairly sharing the available bandwidth. The rapid growing of real-time streaming media applications will bring much UDP traffic without integrating TCP compatible congestion control mechanism into Internet. It threats the quality of service (QoS) of real-time applications and the stability of the current Internet. For this reason, it is desirable to define appropriate rate rule…