延迟的多径路由协议的改进

Ad hoc网络中多径路由算法因其表现出的较好性能受到人们越来越多的关注。但多径路由依旧采用“最小跳数”路由选择机制。很多研究显示最小跳数并不能提供最小的端到端时延保证。采用跨层设计思想,在路由选择机制中引入最短队列长度参数,提出了一种基于延迟的多路径路由算法QAOMDV。仿真结果表明QAOMDV算法可以降低端到端的传输时延,提高数据包的投递率,改善了网络性能。     

Ad Hoc网络中基于延迟的多径路由协议的研究

Ad Hoc网络的频繁变化和节点的移动使得单路径协议的性能受到很大限制.由于多径路由具有稳定和网络资源利用率高的特性,它比原来的单径路由更适合于无线多跳网络.基于跨层设计的思想,提出一种按需多径路由算法QAOMDV.该算法通过节点最短队列长度,把网络层、MAC层和物理层协同起来.在与原有的路由协议的性能进行全面比较的基础上,用NS2仿真工具对改进的算法进行了性能分析和评价,结果表明该改进协议在性能上优于原有路由协议.     

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.     

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。     

基于跨层竞争的同步MAC协议研究

在无线传感器网络(WSNs)中,基于跨层竞争的同步媒体接入控制(MAC)在一周期内可安排多个数据包的多跳传输,传统的协议在同一个数据窗口传输请求数据包和确认数据包,降低了数据窗口的多跳流量的建立,也降低了在多跳场景中的数据包传输率和传输时延性能.本文提出了基于新的基于跨层竞争的同步MAC(CLC-MAC)协议,CLC-MAC协议引用新的周期结构,且其包含两个独立窗口,并由该窗口分别传输数据请求包和确认包,即请求包在数据窗口传输,而确认包在休眠窗口传输.实验数据表明:与先锋路由帧MAC(PRMAC)协议相比,CLC-MAC协议的端到端传输时延和数据包传输率的性能均得到了提高.     

Bandwidth Aggregation-Aware Dynamic QoS Negotiation for Real-Time Video Streaming in Next-Generation Wireless Networks

In next generation wireless networks, Internet service providers (ISPs) are expected to offer services through several wireless technologies (e.g., WLAN, 3G, WiFi, and WiMAX). Thus, mobile computers equipped with multiple interfaces will be able to maintain simultaneous connections with different networks and increase their data communication rates by aggregating the bandwidth available at these networks. To guarantee quality-of-service (QoS) for these applications, this paper proposes a dynamic QoS negotiation scheme that allows users to dynamically negotiate the service levels required for their traffic and to reach them through one or more wireless interfaces. Such bandwidth aggregation (BAG) scheme implies transmission of data belonging to a single application via multiple paths with different characteristics, which may result in an out-of-order delivery of data packets to the receiver and introduce additional delays for packets reordering. The proposed QoS negotiation system aims to ensure the continuity of QoS perceived by mobile users while they are on the move between different access points, and also, a fair use of the network resources. The performance of the proposed dynamic QoS negotiation system is investigated and compared against other schemes. The obtained results demonstrate the outstanding performance of the proposed scheme as it enhances the scalability of the system and minimizes the reordering delay and the associated packet loss rate.     

SE-DSR:一种安全增强的Mesh网络多径动态源路由协议

DSR协议是一种典型的按需多跳路由协议,能够很好地实现既有网络之间的无缝连接又能适应网络的动态变化。为满足无线Mesh网络需求,在分析DSR协议的基础上,本文提出了一种新的安全增强的无线Mesh网络多径DSR路由协议-SE_DSR协议,其基本思想是采用多径路由发现机制为协议提供负载均衡和路由容错能力,采用双向路径信任评估和单向证书链验证的方式为协议提供安全保障。相比传统安全协议,SE_DSR协议虽然牺牲了少量存储开销,但它在保证路由安全的同时缩短了路由发现时延,提高了数据传输速率。性能分析和仿真实验表明,该协议在提供负载均衡和容错的同时还能有效防御多种攻击,具有较高的网络吞吐量和较强的安全性能。     

无线自组网络QoS流的准入控制和流量控制

作者曾提出一个QoS路由和准入控制机制－QDSR,并且在小规模实时视频传输实验床上实现了这些方案。虽然QDSR的准入控制只保证每个节点有足够的可用带宽给准入的所有流使用,但并不保证每个流在较小的时间尺度上得到请求的带宽。为此,作者对QDSR的准入控制机制进行了改进并增加了流量控制机制,以满足实时应用的QoS需求。在NS2仿真环境中实现了QDSR以及对QDSR的改进。仿真结果表明,改进机制改善了QDSR的视频流的传输质量,较好地提高了视频流的吞吐率和投递率,减小了数据包的端到端延迟,并且能很好地满足包括视频传输在内的多种业务的传输需要。     

Multiple sender distributed video streaming

With the explosive growth of video applications over the Internet, many approaches have been proposed to stream video effectively over packet switched, best-effort networks. We propose a receiver-driven protocol for simultaneous video streaming from multiple senders to a single receiver in order to achieve higher throughput, and to increase tolerance to packet loss and delay due to network congestion. Our receiver-driven protocol employs a novel rate allocation algorithm (RAA) and a packet partition algorithm (PPA). The RAA, run at the receiver, determines the sending rate for each sender by taking into account available network bandwidth, channel characteristics, and a prespecified, fixed level of forward error correction, in such a way as to minimize the probability of packet loss. The PPA, run at the senders based on a set of parameters estimated by the receiver, ensures that every packet is sent by one and only one sender, and at the same time, minimizes the startup delay. Using both simulations and Internet experiments, we demonstrate the effectiveness of our protocol in reducing packet loss.     

QoS optimal real-time video streaming in distributed wireless image-sensing platforms

This paper provides a novel real-time video streaming method in distributed wireless image-sensing platforms. It consists of (1) a millimeter-wave (mmW)-based multi-hop routing optimization for real-time video streaming, (2) wireless image-sensing platforms by using the high-efficiency video coding. A mmW wireless communication is a promising technology for increasing capacity in next-generation wireless systems. However, the weakness of mmW signals to (1) do long-distance transmission and (2) survive in non-line-of-sight environments makes the mmW networks need a multi-hop relaying. Thus, this paper focuses on the maximization of video transmission quality of service (QoS) that makes the optimization problem different from the conventional sum-rate maximization. Specifically, this paper develops an algorithm that optimizes the summation of QoS of the individual wireless transmission of different video streams, subject to constraints of separate streams (i.e., minimum requirements in each stream). Experimental results show the proposed approach presents 32 % better performance with 1.84 dB gain in Y-peak signal-to-noise ratio than the widely used max–min flow routing that is generally considered in QoS-sensitive video streaming applications.     

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.     

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.     

On the optimality of field-line routing in massively dense wireless multi-hop networks

We consider the load balancing problem in large wireless multi-hop networks by applying the continuum approximation. The task is to find routes, geometric curves, such that the maximal traffic load in the network is minimized. In finite fixed networks, multi-path routes generally yield a lower congestion and thus allow higher throughput. In contrast, we show that in dense wireless multi-hop networks, the optimal load balancing can be achieved by a destination based single-path routing referred to as field-line routing. This is because any routing can be transformed to the corresponding field-line routing with the same or better performance, by using as paths, the field lines of the so-called destination flow associated with the original routing. The concepts are illustrated with two examples. In the case of a unit disk with unit traffic, the maximal load of 0.389 of a multi-path routing system is reduced to 0.343 by using the field-line routing. Similar improvements are also demonstrated for the unit square.     

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.     

A quality based routing protocol for wireless mesh networks

Wireless mesh networks can provide low-cost solutions for extending the reach of wireless access points by using multi-hop routing over a set of stationary wireless routers. The routing protocol for these networks may need to address quality considerations to meet the requirements of the user. In this paper, we present a quality based routing protocol for wireless mesh networks that tries to maximize the probability of successful transmissions while minimizing the end-to-end delay. The proposed routing protocol uses reactive route discoveries to collect key parameters from candidate routes to estimate the probability of success and delay of data packets transmitted over them. To achieve accurate route quality assessments, a new route quality metric is proposed that uses performance models of data packet transmissions as opposed to estimating route quality from the transmission of control packets, which have different transmission characteristics. These models are developed after careful evaluations of multi-hop wireless transmissions and validated by computer simulations. Relevant parameters that can be used to assess the route quality metric using these models are explained. Extensive performance evaluations of the proposed quality based routing protocol are presented and its benefits in comparison to some other known routing protocols are discussed.     

基于MPEG4的自适应实时流媒体传输

论文阐述了在互联网环境下进行实时流媒体传输中存在的一些难点,并且给出了一个基于MPEG4的实时流媒体传输系统。并且希望能充分利用网络带宽,并且尽力减少媒体流由于网络时延和丢包所产生的影响。同时也关注由于丢包而引起的回放质量的下降问题。     

TCP-ACC: performance and analysis of an active congestion control algorithm for heterogeneous networks

Transmission control protocol (TCP) is a reliable transport layer protocol widely used in the Internet over decades. However, the performances of existing TCP congestion control algorithms degrade severely in modern heterogeneous networks with random packet losses, packet reordering and congestion. In this paper, we propose a novel TCP algorithm named TCP-ACC to handle all three challenges mentioned above. It integrates 1) a real-time reorder metric for calculating the probabilities of unnecessary Fast Retransmit (FRetran) and Timeouts (TO), 2) an improved RTT estimation algorithm giving more weights to packets that are sent (as opposed to received) more recently, and 3) an improved congestion control mechanism based on packet loss and reorder rate measurements. Theoretical analysis demonstrates the equilibrium throughput of TCP-ACC is much higher than traditional TCP, while maintaining good fairness with regard to other TCP algorithms in ideal network conditions. Extensive experimental results using both network emulators and real network show that the algorithm achieves significant throughput improvement in heterogeneous networks as compared with other state-of-the-art algorithms.     

Transmission of layered video streaming via multi-path on ad hoc networks

Mobile ad hoc networks without centralized infrastructure change their topology rapidly because of node mobility, making multimedia applications difficult to run across wireless networks. Moreover, video transmission over ad hoc networks causes frequent transmission loss of video packets owing to end-to-end transmission with a number of wireless links, and requires essential bandwidth and restricted delay to provide quality-guaranteed display. This paper presents an architecture supporting transmission of multiple video streams in ad hoc networks by establishing multiple routing paths to provide extra video coding and transport schemes. This study also proposes an on-demand multicast routing protocol to transport layered video streams. The multicast routing protocol transmits layered video streaming based on a weight criterion, which is derived according to the number of receivers, delay and expiration time of a route. A simulation is performed herein to indicate the viability and performance of the proposed approach. The simulation results demonstrate that the proposed transport scheme is more effective than other video transport schemes with single or multiple paths.

Priority and delay aware packet management framework for real-time video transport over 802.11e WLANs

The rigid delay constraint is one of the most challenging issues in real-time video delivery over wireless networks. The expired video packets will become useless for the decoding and display even if they are received correctly at the receiver. Because the significance of each video packet is different, the schedulers have to take into account not only the urgency of the packet but also its importance in the real-time video applications. However, the existing QoS-based IEEE 802.11e MAC protocol leaves the urgency and the importance of video packets out of consideration. This paper proposes a Priority and Delay Aware Packet Management Framework (PDA-PMF) to improve the transmission quality of real-time video streaming over IEEE 802.11e WLANs. In the MAC layer, this framework estimates the delay of each video packet. Subsequently, video packets are sent or dropped according to both the significance of the video packets and the estimation value of the delay. Simulation results show that the proposed scheme can not only reduce the packet losses, but also protect the more important video packets, so as to improve the received video quality effectively.