VoIP中丢包隐藏技术研究

由于在“尽力型通信”中不可避免的传输错误（如丢包和时延），VoIP的语音质量会潜在地降低。在许多端对端VoIP系统中，语音的服务质量（QoS）很大部分地取决于丢包率和接收端的丢包隐藏算法（PLC）。文中论述了丢包的原因．对当前普遍采用的几种丢包隐藏技术进行了初步分析并进行了比较。     

一种基于VoIP的改进WSOLA丢包隐藏算法

丢包现象严重影响VoIP的通话语音质量。WSOLA(Waveform Similarity Based Overlap-Add)算法是一种基于接收端的丢包隐藏方法,可以较好地提高语音质量。在介绍WSOLA算法原理的基础上,针对该算法中计算互相关系数所需计算量较大,会增加过多计算延时的问题,提出一种互相关系数计算的改进方法。最后通过仿真对重建语音信号质量进行了对比。     

报文分流最优策略研究

在向下一代互联网络演进的过程中,一个重要的趋势是IP网络将成为语音、视频等应用的主要承栽.VoIP是一个重要的语音应用.然而,IP网络的丢包造成了VoIP的服务质量不能得到保证,并且对于VoIP而言,连续丢包对其服务质量的影响要远大于分散丢包.报文分流是近年来学术界讨论的一种提高VoIP服务质量的方法,其基本思想是把1个VoIP会话的报文分散到多个网络链路传输,从而把连续丢包转化为分散丢包,缓解丢包对VoIP服务质量的影响.然而,目前的研究只局限于用一种特定的分流策略(平均分流)说明报文分流的潜力.报文分流的理论基础,比如报文分流能在多大程度上提高VoIP的服务质量,什么是最优的分流策略等,并不明了.对报文分流的理论基础进行了研究,首次给出了分流策略与VoIP服务质量的定量关系描述,给出并证明了Bernoulli网络丢包模型下的最优分流策略.同时,以ns-2仿真实验验证了该最优分流策略在Gilbert网络丢包模型下的有效性.     

基于自适应变速率编码的VoIP网关的研究与应用

针对当前VoIP语音质量易受网络带宽影响的问题，提出了一种基于自适应变速率编码的VoIP网关。通过实时监测RTP语音分组的丢包率，并分析服务质量，自适应地选择最适合当前网络状况的编码速率，以提供一种语音质量和网络状况的最佳组合，从而降低语音分组的丢包率、有效地保障语音服务质量。结合开源项目Asterisk，实现了基于自适应变速率编码的VoIP网关。     

IEEE 802.16系统中自适应的功率节省策略

针对IEEE 802.16系统中基于自适应多速率(AMR)语音编码器的IP语音(VoIP)业务,本文提出了一个自适应的功率节省策略。该策略周期性检测双向会话的语音帧信息,以此来判断上下行业务是否均进入语音静默期,然后自适应地调整功率节省模式参数。从能量节省、丢包率、系统信令开销方面分析了所提策略的性能,并且做了仿真实验。从理论分析和仿真结果可以看出,新策略在保证一定丢包率的基础上,可以比传统策略减少13.4%以上的能量损耗。     

基于E-Model的VoIP语音质量研究

针对目前网络电话语音质量难以准确评价及预测的情况,基于E-Model对VoIP的语音质量进行预测。分析几个主要影响因素,如延时、丢包等对话音质量的影响,构建VoIP语音质量预测模型,将E-Model中未考虑到的抖动因素引入模型公式,着重考虑抖动缓冲区的大小对语音质量的影响。通过设计相关验证实验,证明该模型对VoIP语音质量的预测具有较高的准确度。     

骨干网中 VoIP语音质量的快速评估方法

提出一种基于被动模式流量分析的 VoIP语音质量评估方法 ,以帮助 VoIP运营商对主干网中大量并发 VoIP会话的通话质量进行实时监测。在流量采集的基础上 ,通过流跟踪算法获取 VoIP会话的丢包、时延等基本性能指标 ;通过快速评估模型对 VoIP会话的 R和 MOS值进行计算。实验证明 ,该方法性能较高 ,评估结果准确。     

Vo IP 的语音质量分析与控制

分析了VoIP语音质量的影响因素，通过E模型定量地描述了语音质量与端到端延迟和丢包率的关系。为了控制VoIP的语音质量，计算出VolP系统在各种情况下的语音质量极限，提出一种自适应编码和分组封装的控制策略。将该方法应用于自行开发的IP电话网关，实际测试证明能在很大程度上提高VoIP的语音质量。     

基于E-model的VoIP语音质量评估的研究

为准确评估VoIP）语音质量，对E—model算法进行了深入研究，剖析了E—model算法的组成部分Id，Is，Ie，A，探讨了丢包、延迟和抖动对VoIP质量的影响，并应用该算法对多种语音编码进行了评估。实验证明该客观评估算法主观与客观相关度高，有较强的适应性，可靠性，实用性，完全可用于VoIP语音质量评估。     

一种基于SoPC技术的iLBC语音编解码器设计实现

基于VoIP技术的语音通信发展迅速,单芯片VoIP处理器的设计方法成为当前的研究热点.iLBC作为专为窄带通信而设计的VoIP语音编解码器,可以在丢包率和延迟较高的网络环境中保持良好的语音通话质量,具有广泛的应用前景.传统的基于DSP处理器实现方法具有芯片面积大、功耗高等缺点,难以满足VoIP系统集成度高、低功耗和易于升级等需求.本文提出了一种基于SoPC技术的iLBC语音编解码器实现方案,并对自相关计算算法进行了并行计算硬件IP核设计,提高了系统的集成度、计算性能和可扩展性.理论分析和实验结果表明并行自相关计算结构有效减少了访存次数,可以获得接近30的加速比.     

一种基于G.711包丢失隐藏改进算法的研究

对一种改进的高性能包丢失隐藏算法进行了分析。此算法以G.711脉冲编码调制为基础,建立一种新型线性预测模型,并通过仿真得出此改进算法对VoIP语音丢失包能起到比较好的恢复效果。     

VoIP中一种信包丢失隐藏算法

为减轻基于IP语音（VoIP）网络中因信包丢失而造成的语音失真,提出一种基于双边线性预测和基音调整的信包丢失隐藏算法。该方法利用丢失信包的前一信包或邻接信包（在后一信包可获得的情况下）预测丢失的信包。线性加权经过基音调整后的双边线性预测样点以获得最终的重建信号。最建信号在相位连续性上表现更加合理。经过ITU-T R862协议推荐的PESQ算法测试证明,该算法重建语音信号的质量有了较为明最的改善。     

An Overlay Architecture for High-Quality VoIP Streams

The cost savings and novel features associated with voice over IP (VoIP) are driving its adoption by service providers. Unfortunately, the Internet's best effort service model provides no quality of service guarantees. Because low latency and jitter are the key requirements for supporting high-quality interactive conversations, VoIP applications use UDP to transfer data, thereby subjecting themselves to quality degradations caused by packet loss and network failures. In this paper, we describe an architecture to improve the performance of such VoIP applications. Two protocols are used for localized packet loss recovery and rapid rerouting in the event of network failures. The protocols are deployed on the nodes of an application-level overlay network and require no changes to the underlying infrastructure. Experimental results indicate that the architecture and protocols can be combined to yield voice quality on par with the public switched telephone network     

QoS-aware path switching for VoIP traffic using SCTP

Voice over Internet protocol (VoIP) has been a prevalent multimedia service nowadays. It allows us to transmit voice data over IP networks. However, quality of service (QoS) is a major challenge to VoIP services. It must provide similar quality to traditional public switched telephone network or cellular phone services. Therefore, QoS related protocols have become important for real-time applications. Multi-protocol label switch (MPLS) is one of the important techniques to improve the network performance from QoS point of view. It employs label swapping to speed up packet forwarding. However, when a large number of users utilize VoIP services, the network congestion issue still exists. It causes delay, jitter and packet loss that affect VoIP QoS. In this paper, we propose a QoS-aware path switching strategy by using stream control transmission protocol (SCTP) in MPLS network to improve the VoIP traffic. This was done by employing SCTP selective acknowledgment mechanism to report the transmission parameters of primary path and to determine the criteria to switch to backup path. Simulation results show significant improvement in VoIP QoS.     

Packet loss concealment method based on interpolation in packet voice coding

This paper proposes a packet loss concealment (PLC) method based on redundant information to enhance speech quality under severe network conditions as well as a new model to simulate packet loss effect, end-to-end transmission delay, and jitter on the concealment process. Our proposed method increases the efficiency of the standard ITU-T G.722.2 by improving the quality of the decoded speech under random frame loss conditions over packet-switched networks. The new method achieves a MUltiple Stimuli with Hidden Reference and Anchor (MUSHRA) value exceeding 48.7 at a 20% packet loss ratio (PLR). Numerical analysis indicates that our method outperforms existing alternatives and can be successfully used, even in severe propagation scenarios at the expense of an increase in processing requirements.     

A Garch-based adaptive playout delay algorithm for VoIP

Network delay, packet loss and network delay variability (jitter) are important factors that impact on perceived voice quality in VoIP networks. An adaptive playout buffer is used in a VoIP terminal to overcome jitter. Such a buffer-control must operate a trade-off between the buffer-induced delay and any additional packet loss rate. In this paper, a Garch-based adaptive playout algorithm is proposed which is capable of operating in both inter-talkspurt and intra-talkspurt modes. The proposed new model is based on a Garch model approach; an ARMA model is used to model changes in the mean and the variance. In addition, a parameter estimation procedure is proposed, termed Direct Garch whose cost function is designed to implement a desired packet loss rate whilst minimising the probability of consecutive packet losses occurring. Simulations were carried out to evaluate the performance of the proposed algorithm using recorded VoIP traces. The main result is as follows; given a target Packet Loss Rate (PLR) the Direct Garch algorithm produces parameter estimates which result in a PLR closer than other algorithms. In addition, the proposed Direct Garch algorithm offers the best trade-off between additional buffering delay and Packet Loss Rate (PLR) compared with other traditional algorithms.     

Evaluating perceived voice quality on packet networks using different random neural network architectures

Voice over Internet Protocol (VoIP) is one of the fastest growing technologies in the world. In VoIP speech signals are transmitted over the same network used for data communications. The internet is not a robust network and is subjected to delay, jitter, and packet loss. It is very important to measure and monitor the quality of service (QoS) the users experience in VoIP networks; this is not an easy task and usually requires subjective tests. In this paper we have analyzed three non-intrusive models to measure and monitor voice quality using Random Neural Networks (RNN). A RNN is an open queuing network with positive and negative signals. We have assessed the voice quality based on various parameters i.e. delay, jitter, packet loss, and codec. In our approach we have used the Mean Opinion Score (MOS) calculated using a Perceptual Evaluation of Speech Quality (PESQ) algorithm to generate data for training the RNN model. We have studied two feed-forward models and a recurrent architecture. We have found that the simple feed-forward architecture has produced the most accurate results compared to the other two architectures.     

在WLAN实现VoIP的新兴多路访问协议

VoIP是一种在延时、抖动和丢包率上对WLAN系统要求很严格的实时应用。对于CSMA/CA而言,如果语音和数据传输同时存在的话,想要语音质量满足通话要求,就要在CSMA/CA上增加优先级控制。文章提出了一种新兴的多路访问控制协议,这个协议使基于自治分布式控制的无线局域网能够满足VoIP的要求。     

H.263视频编码流的时域错误掩盖

当H．263编码视频流在Internet上传输时，很容易受到信道错误的影响而丢失数据，由于数据的丢失不但会影响当前帧，还会连续传递到以后的解码帧，而导致图象质量的严重恶化，因此必须采用一定的措施来消除这种影响。目前较常用的错误掩盖算法是时域掩盖算法，而时域掩盖算法是利用参考帧来恢复当前帧损坏的图象数据，其计算较复杂，为此，提出了一种基于块匹配原则的时域掩盖算法，同时用三步搜索代替完全搜索来降低算法的计算复杂性，模拟结果显示，该算法由于能够在很短的处理时间内，获得较好质量的图象，因此能适应于视频会议等实时应用的要求。