无线Mesh网流媒体传输速率控制策略及模型*

基于无线Mesh网（WMN）的网络特征,提出适合WMN流媒体传输的速率控制策略,并给出其相应的模型描述。提出的策略和模型充分利用链路的多样性,降低由重传机制和节点冲突造成的流媒体传输时延增大并改善流媒体传输的性能,同时兼顾WMN接入有线网络的TCP友好性等特征。     

无线实时流媒体传输性能的跨层优化设计

提出一种用于无线实时流媒体传输的优化设计策略,以提高接收方的播放质量。该策略采用跨层设计的方法,利用泊松过程分析链路层数据帧的丢失,同时把链路层最大重传次数映射到端到端时延和丢包率的计算中,自适应地调整MPEG视频帧的发送速率,在视频源数据和冗余数据之间动态分配网络带宽。仿真实验结果表明,该策略能使接收方获得最大的可播放帧率,有效提高流媒体传输的可靠性和实时性。     

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。     

一种无线实时流媒体增强型自适应FEC控制机制

无线网络动态的信道特性和带宽有限等特点,使得在无线环境下为流媒体应用提供QoS保证面临更大的挑战。提出一种用于无线实时流媒体传输的增强型自适应前向纠错控制策略,以提高接收方的播放质量。该策略采用跨层设计的方法,根据当前的网络状态,自适应地调整MPEG视频帧的发送速率,在视频源数据和冗余数据之间动态分配网络带宽。仿真结果表明,该策略能使接收方获得最大的可播放帧率,有效提高流媒体传输的可靠性和实时性。     

Double feedback streaming agent for real-time delivery of media over 3G wireless networks

A network agent located at the junction of wired and wireless networks can provide additional feedback information to streaming media servers to supplement feedbacks from clients. Specifically, it has been shown that feedbacks from the network agent have lower latency, and they can be used in conjunction with client feedbacks to effect proper congestion control. In this work, we propose the double-feedback streaming agent (DFSA) which further allows the detection of discrepancies in the transmission constraints of the wired and wireless networks. By working together with the streaming server and client, DFSA reduces overall packet losses by exploiting the excess capacity Of the path with more capacity. We show how DFSA can be used to support three modes of operation tailored for different delay requirements of streaming applications. Simulation results under high wireless latency show significant improvement of media quality using DFSA over non-agent-based and earlier agent-based streaming systems.     

面向TCP友好性的IP网络实时流媒体流预测控制

为了解决IP网络实时流媒体流控中既要避免网络拥塞又要将接收端缓冲区长度控制在合理范围的难题, 本文提出了一种面向TCP友好性的IP网络实时流媒体流预测控制算法. 该算法将IP网络实时流媒体流控转化为具有TFRC约束时变不确定条件的预测控制问题, 使用改进的动态矩阵控制算法(DMC)进行优化求解. 仿真实验结果表明该方法能够适应IP网络的复杂环境, 满足了TCP友好性要求, 有效地消除了网络传输抖动对播放带来的不利影响, 具有良好的实用性.     

用于流媒体的平滑传输控制协议

为减小传输流媒体业务时的速率抖动,提出一种平滑传输控制协议（Smooth Transmission Control Protocol）。该协议的速率控制主要在接收端实现：接收端不断检测丢包,如果发生丢包,就通过丢包率和当前的发送速率估算端到端的实际传输能力,依据纯属能力减小发送速率;如果没有发生丢包,则模拟TCP的和式增加策略以提高发送速率。使用丢失率降低发送速率,避免了TCP中积式减小的过激行为,减小了速率抖动,获得平滑的发送速率。仿真实验表明,STCP发送速率平滑,时延抖动小,具有TCP友好性,适用于流媒体的传输控制。     

Kalman滤波的自适应链路层FEC控制策略

无线网络动态的信道特性、高误码率和带宽有限等特点,使得在无线环境下为实时流媒体传输提供QoS保证面临更大的挑战。提出一种用于无线实时流媒体传输的自适应链路层FEC控制策略,以显著提高接收方的播放质量。该策略采用跨层设计的方法,基于Kalman滤波器预测当前的网络状态,考虑物理带宽限制和GOP可解码帧数的特性自适应地调整FEC参数N;另一方面,在应用层采用自适应FEC策略,在视频源数据和冗余数据之间动态分配网络带宽。数学分析和仿真验证均表明,该策略能使接收方获得最大的可播放帧率,有效地提高了流媒体传输的可靠性和实时性。     

一种适合无线环境的IPv6头部压缩方法

头部压缩可以提高链路传输效率,但现有压缩方法基于差分编码方式,压缩效率易受丢包影响,在误码率高的无线链路上反而会降低传输效率。为此,提出一种新的IPv6头部压缩方法SBHC。主要思想是利用IPv6头部变化少的特点,采用基于替换的方式实现压缩行为,克服了差分编码方式存在的弊端。性能测试结果表明,SBHC压缩方法可以有效提高无线链路的传输效率。     

无线流媒体主动弃帧策略的仿真研究

仿真研究IEEE802.11g无线网络环境下实时流媒体的性能,在分析和探讨支持实时流媒体应用时无线网络性能瓶颈的基础上,提出一种改进策略--主动弃帧.仿真结果表明,这一策略显著改善了网络性能,为实时流媒体在WLAN上的应用提供更好的服务质量.     

流控制传输协议传输流媒体性能研究

SCTP协议（Stream Control Transmission Protocol）是一种面向报文的、可靠的传输层协议。它基于不可靠的、无连接的分组IP网络,具有多宿（multi-homing）和多流（multi-streaming）等特性。通过仿真的方法分析,证实SCTP在支持实时多媒体网络应用时的许多性能参数明显优于传统的TCP和UDP,包括数据包投递率、带宽开销、端到端延迟、延迟抖动等。在此基础上,探讨一个适合在流媒体应用中,能够有效提高流媒体传输质量的策略。     

Xstream: a middleware for streaming XML contents over wireless environments

XML (extensible Markup Language) has been developed and deployed by domain-specific standardization bodies and commercial companies. Studies have been conducted on a wide variety of issues encompassing XML. In the use of XML for wireless computing, the focus has been on investigating ways to efficiently represent XML data for transmission over a wireless environment. We propose a middleware, Xstream (XML Streaming), for efficiently streaming XML contents over a wireless environment by leveraging the rich semantics and structural characteristics of XML documents and by flexibly managing units containing fragments of data into autonomous units, known as XDU (Xstream Data Unit) fragments. The concept of an XDU is fundamental to the operation of Xstream. It provides for the efficient transfer of documents across a wireless link and allows other issues and challenges pertaining to wireless transmission to be addressed. By fragmenting and organizing an XML document into XDU fragments, we are able to incrementally send fragments across a wireless link, while the receiver is able to perform look-ahead processing of the document without having to wait for the entire document to be downloaded. We propose a fragmenting strategy based on the value of the wireless link's Maximum Transfer Units (MTUs). In addition, we present and evaluate several packetizing strategies, i.e., strategies wherein a collection of XDUs are grouped into a packet to optimize packet delivery and processing. At the receiving end of this process, a reassembly strategy incrementally reconstructs the XML document as XDU fragments are being received, thereby facilitating client application implementation of look-ahead processing.     

Sub-packet forward error correction mechanism for video streaming over wireless networks

Traditional Forward Error Correction (FEC) mechanisms can be divided into Packet level FEC (PFEC) mechanisms and Byte level FEC (BFEC) mechanisms. The PFEC mechanism of recovering from errors in a source packet requires an entire FEC redundant packet even though the error involves a few bit errors. The recovery capability of the BFEC mechanism is only half of the FEC redundancy. Accordingly, an adaptive Sub-Packet FEC (SPFEC) mechanism is proposed in this paper to improve the quality of video streaming data over wireless networks, simultaneously enhancing the recovery performance and reducing the end-to-end delay jitter. The SPFEC mechanism divides a packet into n sub-packets by means of the concept of a virtual packet. The SPFEC mechanism uses a checksum in each sub-packet to identify the position of the error sub-packet. Simulation experiments show the adaptive SPFEC mechanism achieves high recovery performance and low end-to-end delay jitter. The SPFEC mechanism outperforms traditional FEC mechanism in terms of packet loss rate and video Peak Signal-to-Noise Ratio (PSNR). SPFEC offers an alternative for improved efficiency video streaming that will be of interest to the designers of the next generation environments.     

Adaptive hybrid error correction model for video streaming over wireless networks

The Hybrid ARQ (HARQ) mechanism is the well-known error packet recovery solution composed of the Automation Repeat reQuest (ARQ) mechanism and the Forward Error Correction (FEC) mechanism. However, the HARQ mechanism neither retransmits the packet to the receiver in time when the packet cannot be recovered by the FEC scheme nor dynamically adjusts the number of FEC redundant packets according to network conditions. In this paper, the Adaptive Hybrid Error Correction Model (AHECM) is proposed to improve the HARQ mechanism. The AHECM can limit the packet retransmission delay to the most tolerable end-to-end delay. Besides, the AHECM can find the appropriate FEC parameter to avoid network congestion and reduce the number of FEC redundant packets by predicting the effective packet loss rate. Meanwhile, when the end-to-end delay requirement can be met, the AHECM will only retransmit the necessary number of redundant FEC packets to receiver in comparison with legacy HARQ mechanisms. Furthermore, the AHECM can use an Unequal Error Protection to protect important multimedia frames against channel errors of wireless networks. Besides, the AHECM uses the Markov model to estimate the burst bit error condition over wireless networks. The AHECM is evaluated by several metrics such as the effective packet loss rate, the error recovery efficiency, the decodable frame rate, and the peak signal to noise ratio to verify the efficiency in delivering video streaming over wireless networks.     

一种改进的流媒体TFRC拥塞控制算法

为了解决流媒体传输拥塞控制机制的不足,提出了一种基于链路延迟抖动趋势的TFRC改进算法。对传统的TFRC拥塞控制算法以及链路延迟抖动变化趋势进行了分析,采用对链路拥塞状况进行预测的策略,引入抖动因子来修正TFRC的吞吐量公式,由链路延迟抖动的趋势自适应地调整发送速率。仿真实验结果表明,改进算法在保持TCP友好性的前提下,有效提高了流媒体数据传输的平滑性和稳定性。     

Real-time video streaming over multipath in multi-hop wireless networks

Given the limited wireless link throughput, high loss rate, and varying end-to-end delay, supporting video applications in multi-hop wireless networks becomes a challenging task. Path diversity exploits multiple routes for each session simultaneously, which achieves higher aggregated bandwidth and potentially decreases delay and packet loss. Unfortunately, for TCP-based video streaming, naive load splitting often results in inaccurate estimation of round trip time (RTT) and packet reordering. As a result, it can suffer from significant instability or even throughput reduction, which is also validated by our analysis and simulation in multi-hop wireless networks. To make real-time TCP-based streaming viable over multi-hop wireless networks, we propose a novel cross-layer design with a smart traffic split scheme, namely, multiple path retransmission (MPR). MPR differentiates the original data packets and the retransmitted packets and works with a novel QoS-aware multi-path routing protocol, QAOMDV, to distribute them separately. MPR does not suffer from the RTT underestimation and extra packet reordering, which ensures stable throughput improvement over single-path routing. Through extensive simulations, we further demonstrate that, as compared with state-of-the-art multi-path protocols, our MPR with QAOMDV noticeably enhances the TCP streaming throughput and reduces bandwidth fluctuation, with no obvious impact to fairness.     

Error robust scalable audio streaming over wireless IP networks

Streaming high-fidelity audio over wireless Internet protocol (IP) networks is a challenging task because the networks present not only packet losses, but also residual bit errors. These losses and errors have severe adverse effect on the compressed audio bitstream. To solve this problem, this paper introduces error resilience in conjunction with error protection for scalable audio streaming over wireless networks. Specifically, error resilience is achieved by performing bitstream data partitioning and reversible variable length coding in the audio coder. Error protection is provided by layered product channel code to simultaneously handle packet losses and residual bit errors. Both the row and column codes of the product code provide unequal error protection for different layers of the audio bitstream by considering the characteristics of the scalable audio. Rate-distortion optimization is performed to determine the best source-channel coding tradeoff that minimizes the average expected end-to-end distortion. Simulation results demonstrate the effectiveness of our proposed approach.     

流媒体智能调节与控制研究

针对网上教学视频信息传输的需求,对视频传输所涉及的流媒体编码、网络带宽自适应等一些关键技术做了研究。采用两种技术相结合的方法,实现了对流媒体的智能调节和控制,解决了Internet网络环境下远程异步教学信息的传输稳定性等问题。     

多流媒体播放器ActiveX控件开发

多流媒体播放器是接收多流媒体服务器的码流并对其进行解码和控制的软件.目前大部分流媒体播放器只支持一个文件流播放.研究了对多个关联流进行播放控制的播放器ActiveX控件开发,该控件在网络、软件中调用运行效果良好.