共查询到19条相似文献,搜索用时 93 毫秒
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VoIP的语音质量受到丢包、延时、时延抖动等因素影响。为提高语音质量,需要进行丢包补偿处理,如采用WSOLA算法。首先介绍了WSOLA算法的原理,然后提出了用PAOLA对其进行优化的方法,最后对恢复语音质量进行了比较。 相似文献
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基于VoIP队列的主动丢包的调度算法 总被引:1,自引:1,他引:0
讨论了网络拥塞对VoIP应用的影响,基于VoIP应用的特征,提出了一种新的适应于VoIP队列的主动丢包的调度算法——VASA算法。结合VASA和RED算法可实现新的VoIP队列管理和调度模式,从而可以有效改善网络拥塞状态下语音包转发的性能。 相似文献
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IPTV的QoS/QoE受到IP网络丢包的影响是非常大的,丢包问题其实是在IP网络上,特别是在开放Internet环境和无线网络中提供流媒体服务所面临的最大挑战之一。本文从分析丢包现象及其对于IPTV的影响入手,分析比较了几种用于降低丢包造成影响的策略和相关的技术,以及国际标准中在相关领域的最新动态。 相似文献
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基于SIP协议的VoIP系统的QoS机制的研究 总被引:1,自引:0,他引:1
IP电话是一种数字电话,是技术创新的一种通信服务业务,它把语音、压缩编码、打包分组、分配路由、存储交换、解包解压等各种交换处理组合在IP网或互联网上实现语音通信。因特网语音通信是VoIP技术的一个最典型的,也是最有前景的应用领域。介绍了VoIP的通信原理、关键技术和系统通信中的SIP协议,分析了影响VoIP的QoS的主要原因,阐述了时延、抖动、丢包和回声等影响QoS的机理和解决方案。 相似文献
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实时IP 语音通信在数据包会丢失的情况下,语音质量会受到严重影响。为了恢复传输过程中丢失的语音信息,本文提出了一种基于瞬时相位差(Instantaneous Phase Deviation, IPD)和深度神经网络(Deep Neural Network, DNN)的丢包隐藏 (Packet Loss Concealment, PLC)方法。在训练阶段,将语音的对数功率谱(Log Power Spectrum, LPS)和IPD作为训练DNN的输入特征,以学习从接收包到丢失包的映射关系;在重构阶段,将丢包前接收到的语音包送入训练好的DNN中,恢复出丢失包的语音。实验结果表明,在不同丢包率下,所提方法的性能优于传统的基于LPS和DNN的PLC方法。 相似文献
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利用Inernet传输语音,能够提高网络资源的利用率,具有很多优点,但目前Qos还很难保证。本文提出采用联合速率/差错控制模型,同时使用基于发送端的丢包恢复和基于接收端的语音平滑技术,增强了语音抗丢包能力。还采用网络状态反馈技术,使得FEC的冗余方式能够随着网络状态的变化而变化。这个模型不仅能提高服务质量。还能更有效的利用网络资源。 相似文献
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计算机网络节点丢包会严重影响网络通信质量,对相关行业的工作造成较大干扰,因此对丢包节点的检测十分必要。文中对现阶段应用的几种节点检查算法进行了评估和分析,结合既有检测方法存在的问题,提出了基于相似性度量、混合深度学习模型进行丢包节点检测的算法,并通过仿真实验验证了该算法的可行性,以期为相关人员提供参考。 相似文献
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对LRMP中的局部错误恢复技术进行了研究,为适应实时音频传输的需要提出了改进方法。通过对层次划分原则的改进可缩短恢复时间。对执行恢复条件的判断可保证接收方在遭遇严重的连续丢包时也能恢复。 相似文献
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An important issue for video transmission over IP networks is the preservation of perceived video quality despite packet loss. Packet loss can be detrimental to compressed video. However, reducing packet loss to a very low level is difficult with current techniques. Furthermore, even a very low objective loss probability can still seriously distort perceived video quality. This paper presents a packet scheduling scheme at a router which addresses the loss issues of networked video. Experiments using real video data show that the proposed scheme can significantly improve the visual quality of video and network efficiency. Moreover, it can provide different classes of videos with different levels of loss guarantees, while maintaining service fairness among the same class of videos. 相似文献
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A survey of packet loss recovery techniques for streaming audio 总被引:4,自引:0,他引:4
We survey a number of packet loss recovery techniques for streaming audio applications operating using IP multicast. We begin with a discussion of the loss and delay characteristics of an IP multicast channel, and from this show the need for packet loss recovery. Recovery techniques may be divided into two classes: sender- and receiver-based. We compare and contrast several sender-based recovery schemes: forward error correction (both media-specific and media-independent), interleaving, and retransmission. In addition, a number of error concealment schemes are discussed. We conclude with a series of recommendations for repair schemes to be used based on application requirements and network conditions 相似文献
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Improving protocol robustness in ad hoc networks through cooperative packet caching and shortest multipath routing 总被引:1,自引:0,他引:1
A mobile ad hoc network is an autonomous system of infrastructure-less, multihop, wireless mobile nodes. Reactive routing protocols perform well in this environment due to their ability to cope quickly against topological changes. This paper proposes a new routing protocol named CHAMP (caching and multiple path) routing protocol. CHAMP uses cooperative packet caching and shortest multipath routing to reduce packet loss due to frequent route failures. We show through extensive simulation results that these two techniques yield significant improvement in terms of packet delivery, end-to-end delay and routing overhead. We also show that existing protocol optimizations employed to reduce packet loss due to frequent route failures, namely local repair in AODV and packet salvaging in DSR, are not effective at high mobility rates and high network traffic. 相似文献
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An important issue for video transmission over IP networks is preservation of perceived video quality despite packet loss. Packet loss can be detrimental to compressed video. However, reducing packet loss to a very low level is difficult with current loss control techniques. Furthermore, even a very low objective loss probability can still seriously distort perceived video quality. This paper presents two buffer management schemes using video characteristics. They increase maximum loss tolerance for a desired level of video quality, providing better quality at equal loss ratios, when compared to a conventional buffer management scheme. Meanwhile, service fairness and network efficiency are also improved. 相似文献
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End-to-end Internet packet dynamics 总被引:2,自引:0,他引:2
We discuss findings from a large-scale study of Internet packet dynamics conducted by tracing 20000 TCP bulk transfers between 35 Internet sites. Because we traced each 100-kbyte transfer at both the sender and the receiver, the measurements allow us to distinguish between the end-to-end behavior due to the different directions of the Internet paths, which often exhibit asymmetries. We: (1) characterize the prevalence of unusual network events such as out-of-order delivery and packet replication; (2) discuss a robust receiver-based algorithm for estimating “bottleneck bandwidth” that addresses deficiencies discovered in techniques based on “packet pair;” (3) investigate patterns of packet loss, finding that loss events are not well modeled as independent and, furthermore, that the distribution of the duration of loss events exhibits infinite variance; and (4) analyze variations in packet transit delays as indicators of congestion periods, finding that congestion periods also span a wide range of time scales 相似文献
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Raed T. Al-Zubi Marwan Krunz Ghazi Al-Sukkar Mohammed Hawa Khalid A. Darabkh 《Wireless Personal Communications》2014,75(2):943-963
Most of the schemes that were proposed to improve the performance of transmission control protocol (TCP) over mobile ad hoc networks (MANETs) are based on a feedback from the network, which can be expensive (require extra bandwidth) and unreliable. Moreover, most of these schemes consider only one cause of packet loss. They also resume operation based on the same stand-by parameters that might vary in the new route. Therefore, we propose two techniques for improving the performance of TCP over MANETs. The first one, called TCP with packet recycling (TCP-PR), allows the nodes to recycle the packets instead of dropping them after reaching the retransmission limit at the MAC layer. In the second technique, which is called TCP with adaptive delay window (TCP-ADW), the receiver delays sending TCP ACK for a certain time that is dynamically changed according to the congestion window and the trip time of the received packet. TCP-PR and TCP-ADW are simple, easy to implement, do not require network feedback, compatible with the standard TCP, and do not require distinguishing between the causes of packet loss. Our thorough simulations show that the integration of our two techniques improves the performance of TCP over MANETs. 相似文献
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《Signal Processing: Image Communication》2001,16(8):817-825
To efficiently combat the signal loss of MPEG-2 transmission over unreliable networks, priority encoding transmission, unequal packet loss protection and priority dropping techniques have been studied in many papers. Those studies are based on the qualitative analysis of different importance of signals, without quantitative investigation of signal loss effect on video quality. In this paper, MPEG-2 packet loss effect on video quality is quantitatively investigated, a temporal layered signal model is described and evaluated, a quality measure for reconstructed pictures called macroblock impairment ratio is suggested and defined. The investigation and the model are specified for MPEG-2, but the principles and the methods are suitable for any layered video. These are useful for the development of efficient schemes and protocols for packet video transmission over unreliable networks. 相似文献
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The BLUE active queue management algorithms 总被引:6,自引:0,他引:6
Wu-chang Feng Shin K.G. Kandlur D.D. Saha D. 《Networking, IEEE/ACM Transactions on》2002,10(4):513-528
In order to stem the increasing packet loss rates caused by an exponential increase in network traffic, the IETF has been considering the deployment of active queue management techniques such as RED (random early detection) (see Floyd, S. and Jacobson, V., IEEE/ACM Trans. Networking, vol.1, p.397-413, 1993). While active queue management can potentially reduce packet loss rates in the Internet, we show that current techniques are ineffective in preventing high loss rates. The inherent problem with these algorithms is that they use queue lengths as the indicator of the severity of congestion. In light of this observation, a fundamentally different active queue management algorithm, called BLUE, is proposed, implemented and evaluated. BLUE uses packet loss and link idle events to manage congestion. Using both simulation and controlled experiments, BLUE is shown to perform significantly better than RED, both in terms of packet loss rates and buffer size requirements in the network. As an extension to BLUE, a novel technique based on Bloom filters (see Bloom, B., Commun. ACM, vol.13, no.7, p.422-6, 1970) is described for enforcing fairness among a large number of flows. In particular, we propose and evaluate stochastic fair BLUE (SFB), a queue management algorithm which can identify and rate-limit nonresponsive flows using a very small amount of state information. 相似文献
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Vinod M. Vokkarane 《Optical Switching and Networking》2007,4(1):20-32
In this paper, we discuss different signaling techniques for optical burst-switched networks. We develop a generalized signaling framework for optical burst-switched networks, which provides guidelines about the performance of each signaling technique based on the different parameters in the framework. The two commonly used signaling techniques in optical burst switching are two-way based tell-and-wait (TAW) and one-way based just-enough-time (JET). TAW suffers from high end-to-end packet delay, while JET suffers from high packet loss. There is no signaling technique that offers flexibility in terms of both loss and delay. We propose a hybrid signaling technique called intermediate-node-initiated (INI) signaling for optical burst-switched networks. INI can provide different levels of loss and delay characteristics based on end-user application requirements. The granularity of INI ranges between the one-way based and the two-way based signaling techniques. In INI reservation of channels is initiated at an intermediate node, known as the initiating node, in both forward and backward directions at the same time. We show that by appropriately selecting the initiating node, we can simulate both TAW and JET using the INI signaling technique. Through simulations, we shown that INI performs better than TAW in terms of average end-to-end packet delay and better than JET in terms of burst loss probability. We extend the INI signaling technique to provide QoS differentiation in the OBS core, differentiated INI (DINI), by carefully choosing different initiation nodes depending on delay and loss requirements of end-user applications. Through extensive simulations, we show that the DINI technique outperforms the existing offset-based QoS technique. 相似文献