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.     

基于Kalman滤波的无线流媒体自适应混合FEC/ ARQ控制策略

提出了一种用于无线实时流媒体传输的数据链路层自适应混合FEC/ARQ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,自适应地调整FEC参数N和ARQ参数Nmax;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。     

Multiple TFRC Connections Based Rate Control for Wireless Networks

Rate control is an important issue in video streaming applications for both wired and wireless networks. A widely accepted rate control method in wired networks is equation based rate control , in which the TCP friendly rate is determined as a function of packet loss rate, round trip time and packet size. This approach, also known as TCP friendly rate control (TFRC), assumes that packet loss in wired networks is primarily due to congestion, and as such is not applicable to wireless networks in which the bulk of packet loss is due to error at the physical layer. In this paper, we propose multiple TFRC connections as an end-to-end rate control solution for wireless video streaming. We show that this approach not only avoids modifications to the network infrastructure or network protocol, but also results in full utilization of the wireless channel. NS-2 simulations, actual experiments over 1$times$RTT CDMA wireless data network, and and video streaming simulations using traces from the actual experiments, are carried out to validate, and characterize the performance of our proposed approach.     

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。     

Evaluating the significance of error checksums for wireless video streaming

Efficient streaming of bandwidth intensive and delay sensitive multimedia contents over error prone wireless links has proven to be one of the most challenging problems of current era of digital communication. Applying unequal error protection strategies and avoiding unnecessary packet discard at various network levels yield valuable outcomes. In this article, we have proposed the idea of discriminating classified video streaming calls from the data packeting over IEEE WLAN through bit demarcation in network packet headers. Error computation at various network levels are evaluated and disabled in order to attain increased throughput characterized by the higher number of packets available for decoding, enhanced multimedia visual quality due to gap elimination (appears as a consequence of some frame loss), efficient utilization of link bandwidth with no re-transmissions and reduced delays with least error checksum computations and packet re-transmissions. Moreover, collaborative estimation of various layers parameters results in proficient selection of streaming parameters like group of picture structure, inter spacing of anchor frames, constellation coding and signal power. The proposed system will be helpful in future information and communication systems by providing reliable video streaming over wireless.     

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.     

无线网络中基于流媒体传输的自适应TFRC机制

在无线网络高误码率的环境下, 经典TFRC机制会将无线误码丢包误认为拥塞丢包, 导致吞吐量过度降低. 针对无线网络实时流媒体业务的传输控制问题, 提出了一种改进型动态自适应TFRC机制(Adaptive-TFRC). 它在接收端利用丢包区分参数来真实反映网络的状态(即拥塞或者误码), 然后反馈至发送端, 同时对经典TFRC机制的吞吐量模型公式进行改进, 最终能够根据实时网络条件动态自适应地调节传输速率. 仿真结果表明, Adaptive-TFRC机制能够有效地提高网络吞吐量, 降低实时业务流的延时抖动, 同时能够进一步改善TCP业务的友好性传输, 从而保证无线网络实时流媒体的服务质量.     

高带宽无线局域网中PI速率控制算法

针对传统传输控制协议（TCP）在高带宽、无线网络中性能表现不佳的问题, 建立了一个端到端的虚拟路由器模型, 提出了一个端到端的发送端比例积分（PI）速率控制算法。根据RTT的变化, 利用PI控制器计算发送端的发送速率, 使瓶颈节点的队长稳定在一个目标位置, 减少拥塞丢包, 避免无线链路错误丢包引起的对拥塞窗口的错误调整。仿真结果表明, 与传统拥塞控制协议相比, 新机制能较好地控制路由器队长、提高网络负载的稳定性和网络吞吐率。     

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.     

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。     

A framework of cross-layer design for multiple video streams in wireless mesh networks

In this paper, a novel cross-layer design framework for multiple realtime video traffics in CDMA wireless mesh networks is proposed. First, the performances of application, physical, MAC, and network layers are modeled by some classical models under reasonable assumptions. Then, we present a framework in which source coding, power control, ARQ control, and delay partitioning functionalities at different layers can be jointly optimized. Our objective is to maximize the video quality under strict end-to-end delay constraints through adjusting source coding rate, end-to-end delay distribution, and each node’s transmit power. This optimization problem is proved to be a nonlinear but log-convex one. Finally, we propose a centralized solution based on the classical convex programming method, as well as a partially distributed solution based on the Lagrangian dual decomposition technique. The both solutions are proved to converge to the global optimum of the above problem.     

An improved UDP protocol for video transmission overInternet-to-wireless networks

Packet video will become a significant portion of emerging and future wireless/Internet traffic. However, network congestion and wireless channel error yields tremendous packet loss and degraded video quality. In this paper, we propose a new complete user datagram protocol (CUDP), which utilizes channel error information obtained from the physical and link layers to assist error recovery at the packet level. We propose several maximal distance separable (MDS) code-based packet level error control coding schemes and derive analytical formulas to estimate the equivalent video frame loss for different versions of user datagram protocol (UDP). We validate the proposed packet coding and CUDP protocol using MPEG-coded video under various Internet packet loss and wireless channel profiles. Theoretic and simulation results show that the video quality can be substantially improved by utilizing the frame error information at UDP and application layer     

基于GM(1,1)预测的MANET拥塞控制

提出一种用于无线移动Ad hoc网络的TCP自适应拥塞控制机制(TCP_Acc),在给定数据链路层统计带宽的情况下,使用GM(1,1)模型预测未来的网络状态,并根据预测得到的带宽信息自适应地调节拥塞窗口。仿真实验结果表明,该机制能够有效地改进无线移动Ad hoc网络中实时TCP通信的可靠性和无线实时通信的服务质量,如较低的丢包率以及端到端延迟等。     

A Control-Theoretic Approach to Rate Control for Streaming Videos

As streaming videos are becoming increasingly popular, it is important to understand the end-to-end streaming system and to develop effective algorithms for quality control. In this paper, we address the problem of rate control for streaming videos with a control-theoretic approach. Among the various control knobs, video bit rate is one of the most effective in the sense that it has a direct impact on the interaction between the video coder and network system. While increasing rate reduces the coder-induced distortion, it may also cause congestion at a bottleneck link. The packet loss due to congestion will, then, increase the distortion of the decoded video. We model end-to-end video steaming as a feedback control system, taking into account video codec and sequence characteristics, rate control, active queue management, and receiver feedback. We then develop effective proportional (P) controllers to stabilize the received video quality as well as the bottleneck link queue, for both homogeneous and heterogeneous video systems. Simulation results are presented to demonstrate the efficacy of the P controllers and the viability of the proposed control-theoretic approach.     

Providing adaptive QoS to layered video over wireless local area networks through real-time retry limit adaptation

Robust streaming of video over 802.11 wireless LANs (WLANs) poses many challenges, including coping with packets losses caused by network buffer overflow or link erasures. In this paper, we propose a novel error protection method that can provide adaptive quality-of-service (QoS) to layered coded video by utilizing priority queueing at the network layer and retry-limit adaptation at the link layer. The design of our method is motivated by the observation that the retry limit settings of the MAC layer can be optimized in such a way that the overall packet losses that are caused by either link erasure or buffer overflow are minimized. We developed a real-time retry limit adaptation algorithm to trace the optimal retry limit for both the single-queue (or single-layer) and multiqueue (or multilayer) cases. The video layers are unequally protected over the wireless link by the MAC with different retry limits. In our proposed transmission framework, these retry limits are dynamically adapted depending on the wireless channel conditions and traffic characteristics. Furthermore, the proposed priority queueing discipline is enhanced with packet filtering and purging functionalities that can significantly save bandwidth by discarding obsolete or un-decodable packets from the buffer. Simulations show that the proposed cross-layer protection mechanism can significantly improve the received video quality.     

一种改进的瑞利小波算法无线传感器网络流媒体传输研究

针对无线传感器网络中带宽和能量受限、误码率高、信道不稳定等因素严重影响了实时流媒体传输的问题,采用瑞利小波模型模拟无线传感器网络流媒体通信,并给出概率分布和突发特性的分析模型,基于Kal-man滤波器实时预测网络带宽,自适应地在SCTP与PRSCTP之间进行切换。仿真实验表明,瑞利小波模型能够准确地描述实时流媒体通信流,Kalman滤波器可以准确地预测实时网络带宽,而且基于带宽预测的流媒体传输技术与原有的技术相比在分组成功投递率、端到端时延和吞吐率上均具有良好的性能。     

基于链路状态感知的无线Mesh网优化路由协议

在分析无线Mesh网路由协议所面临的挑战的基础上,结合无线Mesh网络的性能要求,以OLSR协议为原 型,采用跨层设计理论,提出了一种基于链路状态良好程度的路由协议工R-)工SR。该协议引入了认知无线网络中的 环境感知推理思想,通过对节点负载、链路投递率和链路可用性等信息进行感知,并以此为依据对链路质量进行推理, 将其作为路由选择的依据,实现对路由的优化选择,提高网络的吞吐量,达到负载均衡。仿真结果表明,工R-OI_SR能 够在很大程度上提高网络中分组的递交率,降低平均端到端时延,在一定程度上达到负载均衡。     

A robust method for soft IP handover

Multimedia streaming over wireless networks - often called mobile multimedia streaming lets users access music, movie, and news services at any time, regardless of location. Given that multimedia streaming is a key goal of third-generation and future wireless networks, vendors will soon deploy streaming clients in advanced mobile terminals. Current mobile terminals, however, fail to adequately support mobile multimedia communication because wireless networks have high packet-loss rates. To eliminate packet loss during handover, we use a packet path diversity scheme and an end-to-end bicasting mechanism that enables soft IP handover. To offset wireless errors, we use a forward error correction (FEC) scheme and embed it in the bicasting mechanism. Our bicasting method encodes the data stream and then splits it, providing more effective diversity than general bicasting, which sends the same data down both paths.' To support our method, we propose the mobile multimedia streaming protocol (MMSP), a new transport-layer protocol that supports multihoming and bicasting in combination with FEC.     

无线上行视频流的跨层容错传输方法

如何通过资源受限的移动通信终端提升无线上行视频流的抗误性能是亟待解决的重要问题。通过不同通信层次的联合调度,提出了一种跨层容错传输方案。移动通信终端的网络层代理首先利用容错包调度为视频流的延时约束帧集合提供重要性分类,随后该终端的链路层代理利用无线链路单元的优先级调度实现选择性重传。在调度延时与传输带宽限制下,跨层容错传输能够将突发错误转移到延时约束帧集合的低优先级视频数据中,从而在突发易错传输环境中实现了无线链路单元粒度的渐进式传输和平稳退化。     

Dynamical combination of byte level and Sub-Packet level FEC in HARQ mechanism to reduce error recovery overhead on video streaming over wireless networks

Byte level Forward Error Correction (B-FEC) is efficient for recovery from uniform bit errors, but not suitable to handle recovery from burst bit errors. Conversely, Sub-Packet level Forward Error Correction (SP-FEC) can alleviate the problem of large encoding/decoding delay jitter in Packet level Forward Error Correction (P-FEC) to efficiently handle recovery from burst bit errors, but has large error recovery overhead as P-FEC for recovery from uniform bit errors. This paper proposes a dynamic combination of byte level and Sub-Packet level Forward Error Correction (BSP-FEC) in the Hybrid Automatic Repeat reQuest (HARQ) mechanism to reduce the error recovery overhead. BSP-FEC not only can solve the problems appearing in B-FEC and SP-FEC, but also can get the advantages of B-FEC and SP-FEC in the HARQ mechanism. BSP-FEC replaces the SP-FEC checksum with B-FEC and uses Automatic Repeat reQuest (ARQ) when the network condition deteriorates. BSP-FEC not only utilizes an overhead cost model to dynamically decide the SP-FEC parameter and the B-FEC bit rate according to network conditions, but also utilizes a time constraint model to decide the ARQ retry limit. BSP-FEC dynamically adjusts the FEC redundancy to save bandwidth and improves the Decodable Frame Rate (DFR) and the Peak Signal to Noise Ratio (PSNR) of the delivered video streaming. Accordingly, BSP-FEC can improve multimedia communication performance to both avoid network congestion and shorten end-to-end delay by decreasing effective packet loss rate and packet recovery overhead. Because of the low packet recovery overhead, furthermore, BSP-FEC allows applications to transmit more application data in networks with limited bandwidth. Considering the compatibility, BSP-FEC is implemented in the application layer as the end-to-end protection method to protect packets from errors in wired/wireless networks. Numerical and simulation experimental results show that BSP-FEC obtains better recovery efficiency with the minimum error recovery overhead.