突发大时延下的IP语音缓冲算法

当网络中存在突发大时延时,就会出现极大丢包率或极大端到端时延的情况,从而难以获得好的语音质量。对于这个问题,提出针对突发大时延存在下的自适应语音缓冲算法。算法通过控制语音包在语音缓冲队列中的位置来控制语音包的播放时间,从而可以尽量减小语音裂缝(Gap)的出现。算法将突发大时延存在下的最大丢包率可以扩大到20％,而一般的预测算法只能容忍5-10％的最大丢包率。通过基于听觉模型的客观音质评价(PESQ)仿真计算,以及实际应用表明本文的算法对有突发大时延存在的网络中的语音通信质量有一定的改善作用。     

IP语音包的自适应编码和封装算法的研究

IP电话与传统电话相比语音质量较差，其中最主要的原因是因特网的带宽变化较大，导致丢包率较大。该文根据因特网带宽变化的特点提出了1种应用在IP电话网关中的语音自适应编码与封装策略，采用该策略的编码器能根据网络的带宽变化动态调节语音编码速率和语音包封装大小。据此，本文提出了4种算法：一种基于RTP协议语音包丢失率的计算算法、变速率编码算法，不同长度IP语音包的封装算法和根据丢包率来调整编码速率和封装的自适应算法。     

TD-LTE系统VoLTE语音质量影响因素及提升措施分析

随着VoLTE语音业务的规模发展以及TD-LTE网络质量的不尽完善,VoLTE语音质量受到一定影响。本文分析了TD-LTE无线网络中影响VoLTE语音质量的主要因素,包括频繁切换、干扰、丢包;然后对无线基站侧功能进行分析,梳理出能够提升VoLTE语音质量的措施。     

一种适应移动设备的CELP宽带语音编码算法

本文介绍了一种新的CELP宽带语音编码算法,由于使用了分带技术,使得该算法同时支持窄带和宽带等不同采样率的语音编码.该算法还有其它算法所不具有的可变码率、平均码率控制、语音激活和非连续发送等功能,所以在保证语音质量的同时码率也比较低.因为在编码时尽量避免了利用其它帧的信息,所以对丢包有一定的鲁棒性,再加上计算复杂度不高,延时小,所以该算法适合于应用在嵌入式的移动设备上.     

基于NS的自适应多速率网络语音通信的性能分析

目前,几乎所有的语音电话系统（VoIP）都采用固定速率传输,这使得网络丢包,特别是连续丢包无法避免,因此导致了严重的语音质量下降.针对这一问题,给出了一种新的抗分组丢失的网络语音通信系统,并用网络仿真软件NS（network simulator）对该系统进行了性能分析,仿真实验证明,所提出的网络语音通信系统在网络丢包、平均延迟和主观听觉方面明显优于传统的IP语音电话系统.     

基于延迟快速调度算法的Vo LTE低丢包评估

为了改善VoLTE语音通话时的用户感知,提升VoLTE通话的语音质量,参考延迟快速调度算法,从承载VoLTE语音的相关网元入手,分析其主要功能和相关参数,提出VoLTE语音低丢包问题优化评估研究,建立参数模型,选出最优参数并推广。     

VoLTE高丢包问题分层优化方法研究

VoLTE是LTE网络语音演进的最终目标方案。VoLTE语音业务中,丢包问题是影响语音质量的关键因素之一。高丢包会导致语音吞字、断续、单通等问题,严重影响用户感知。本文从VoLTE协议栈入手,分析各协议层主要功能,提出高丢包优化方法,展示参数试点效果。     

一种基于RTP减少包丢失的控制算法及其在VoIP中的应用

本文首先考虑丢包率的控制方案，然后讨论VoIP网关对丢包率的监测，最后在简单带宽调整算法的基础上提出一种改进的控制算法——传输速率自适应调整算法，并将算法运用到VoIP系统中，提出一种自适应变速率语音编码器，来控制语音包的丢失。     

一种新的语音流队列管理调度机制

本文提出一种新的语音流队列管理调度机制,结合随机早期探测(RED)和主动丢包调度算法实现因特网语音流的队列管理和调度.采用仿真方法分析了新机制的性能特征,并与RED做了性能对比.当网络拥塞时,该算法可有效改善包转发的性能.语音质量测试表明新机制是可行的和有效的.     

一种新的语音通信抗分组丢失方法—分布式子帧交织描述

实时语音经不可靠网络传输时会遭遇丢包现象,导致语音质量的下降。提出了个新的分布式子帧交织方案,用于多描述语音编码中。它在两个相邻的语音帧间进行分布式子帧交织,以生成两个相关的描述;每一个描述能提供可接受的语音质量。文中方案的性能通过客观音质评价指标PESQ来评估,实验结果证明本方案适用于不可靠和带宽有限的分组网络,发生丢包时可避免重传,并保证可接受的语音质量。     

Variable bit-rate coding of video signals for ATM networks

Statistical characteristics of video signals for video packet coding, are clarified and a variable-bit-rate coding method for asynchronous transfer mode (ATM) networks is described that is capable of compensating for packet loss. ATM capabilities are shown to be greatly affected by delay, delay jitter, and packet loss probability. Packet loss has the greatest influence on picture quality. Packets may be lost either due to random bit error in a cell header or to network control when traffic is congested. A layered coding technique using discrete-cosine transform (DCT) coding is presented which is suitable for packet loss compensation. The influence of packet loss on picture quality is discussed, and decoded pictures with packet loss are shown. The proposed algorithm was verified by computer simulations     

无感知分组丢失下的无线传感器网络链路质量评估模型

针对现有链路质量评估方法没有考虑到不完整分组对链路产生的影响,将无线传感器网络的链路分组丢失分为感知分组丢失和无感知分组丢失,分析了无感知分组丢失产生的原因,在对无感知分组丢失进行度量的基础上,提出一种无感知分组丢失下的链路质量评估模型。该模型采用卡尔曼滤波对获取的CCI进行降噪处理,基于Cubic模型、最小二乘法拟合出CCI与无感知分组丢失率的关系模型。实验结果表明,无感知分组丢失下的链路质量评估模型是合理的,通过该模型求得的无感知分组丢失率与实测值接近。     

Single-ended parametric voicing-aware models for live assessment of packetized VoIP conversations

The perceptual quality of VoIP conversations depends tightly on the pattern of packet losses, i.e., the distribution and duration of packet loss runs. The wider (resp. smaller) the inter-loss gap (resp. loss gap) duration, the lower is the quality degradation. Moreover, a set of speech sequences impaired using an identical packet loss pattern results in a different degree of perceptual quality degradation because dropped voice packets have unequal impact on the perceived quality. Therefore, we consider the voicing feature of speech wave included in lost packets in addition to packet loss pattern to estimate speech quality scores. We distinguish between voiced, unvoiced, and silence packets. This enables to achieve better correlation and accuracy between human-based subjective and machine-calculated objective scores.     

An optical packet contention resolution supporting QoS differentiation with LRWC wavelength converters for OPS

The wavelength conversion is regarded as an effective way to resolve the optical packet contention in the wavelength domain for optical packet switching. An optical packet switching node, based on shared-per-node equipped with limited range wavelength converters and parametric wavelength converters (SPN-LP), is designed to further reduce optical packet loss probability. A novel optical packet contention resolution with priority differentiation wavelength scheduling algorithm to support quality of service (QoS) for the SPN-LP architecture is put forward in the article. The simulation results show that proposed optical packet resolution enables a good QoS differentiation, namely the high priority contending optical packet has the sufficient low packet loss probability.     

Improving the R-score of an adaptive VoIP codec in IEEE 802.16 networks

Worldwide Interoperability for Microwave Access (WiMAX) technology, which is based on the IEEE 802.16 standard, supports different quality of service (QoS) for different services. WiMAX is expected to support QoS in real-time applications such as Voice over Internet Protocol (VoIP). When network congestion occurs, the VoIP bit rate needs to be adjusted to achieve the best speech quality. In this study, we propose a new scheme called Adaptive VoIP Level Coding (AVLC). This scheme takes into consideration network conditions (packet delay and packet loss) and a connection’s modulation scheme. The amount of data that can be transmitted increases with the speed of the modulation scheme. When network congestion occurs, AVLC scheme prioritizes reducing the bit rate of a connection that has a slower modulation scheme to mitigate congestion. Depending on network conditions, such as modulation scheme, packet delay, packet loss, and residual time slot, we use the G.722.2 codec to adjust each connection’s bit rate. Simulations are conducted to test the performance (network delay, packet loss, number of modulation symbols, and R-score) of the proposed scheme. The simulation results indicate that speech quality is improved by the use of AVLC.     

Adaptive Playout Buffer Algorithm for Enhancing Perceived Quality of Streaming Applications

An end-to-end packet delay in the Internet is an important performance parameter, because it heavily affects the quality of real-time applications. In the current Internet, however, because the packet transmission qualities (e.g., transmission delays, jitters, packet losses) may vary dynamically, it is not easy to handle a real-time traffic. In UDP-based real-time applications, a smoothing buffer (playout buffer) is typically used at a client host to compensate for variable delays. The issue of playout control has been studied by some previous works, and several algorithms controlling the playout buffer have been proposed. These studies have controlled the network parameters (e.g., packet loss ratio and playout delay), not considered the quality perceived by users. In this paper, we first clarify the relationship between Mean Opinion Score (MOS) of played audio and network parameters (e.g., packet loss, packet transmission delay, transmission rate). Next, utilizing the MOS function, we propose a new playout buffer algorithm considering user's perceived quality of real-time applications. Our simulation and implementation tests show that it can enhance the perceived quality, compared with existing algorithms.     

Evaluation of temporal variation of video quality in packet loss networks

We examine the effect that variations in the temporal quality of videos have on global video quality. We also propose a general framework for constructing temporal video quality assessment (QA) algorithms that seek to assess transient temporal errors, such as packet losses. The proposed framework modifies simple frame-based quality assessment algorithms by incorporating a temporal quality variance factor. We use packet loss from channel errors as a specific study of practical significance. Using the PSNR and the SSIM index as exemplars, we are able to show that the new video QA algorithms are highly responsive to packet loss errors.     

Adaptive Speech Streaming Based on Packet Loss Prediction Using Support Vector Machine for Software‐Based Multipoint Control Unit over IP Networks

An adaptive speech streaming method to improve the perceived speech quality of a software‐based multipoint control unit (SW‐based MCU) over IP networks is proposed. First, the proposed method predicts whether the speech packet to be transmitted is lost. To this end, the proposed method learns the pattern of packet losses in the IP network, and then predicts the loss of the packet to be transmitted over that IP network. The proposed method classifies the speech signal into different classes of silence, unvoiced, speech onset, or voiced frame. Based on the results of packet loss prediction and speech classification, the proposed method determines the proper amount and bitrate of redundant speech data (RSD) that are sent with primary speech data (PSD) in order to assist the speech decoder to restore the speech signals of lost packets. Specifically, when a packet is predicted to be lost, the amount and bitrate of the RSD must be increased through a reduction in the bitrate of the PSD. The effectiveness of the proposed method for learning the packet loss pattern and assigning a different speech coding rate is then demonstrated using a support vector machine and adaptive multirate‐narrowband, respectively. The results show that as compared with conventional methods that restore lost speech signals, the proposed method remarkably improves the perceived speech quality of an SW‐based MCU under various packet loss conditions in an IP network.     

A novel adaptive traffic prediction AQM algorithm

In the Internet, network congestion is becoming an intractable problem. Congestion results in longer delay, drastic jitter and excessive packet losses. As a result, quality of service (QoS) of networks deteriorates, and then the quality of experience (QoE) perceived by end users will not be satisfied. As a powerful supplement of transport layer (i.e. TCP) congestion control, active queue management (AQM) compensates the deficiency of TCP in congestion control. In this paper, a novel adaptive traffic prediction AQM (ATPAQM) algorithm is proposed. ATPAQM operates in two granularities. In coarse granularity, on one hand, it adopts an improved Kalman filtering model to predict traffic; on the other hand, it calculates average packet loss ratio (PLR) every prediction interval. In fine granularity, upon receiving a packet, it regulates packet dropping probability according to the calculated average PLR. Simulation results show that ATPAQM algorithm outperforms other algorithms in queue stability, packet loss ratio and link utilization.     

CCI-based link quality estimation mechanism for wireless sensor networks under non-perceived packet loss

This paper proposes a chip correlation indicator (CCI)-based link quality estimation mechanism for wireless sensor networks under non-perceived packet loss. On the basis of analyzing all related factors, it can be concluded that signal-to-noise rate (SNR) is the main factor causing the non-perceived packet loss. In this paper, the relationship model between CCI and non-perceived packet loss rate (NPLR) is established from related models such as SNR versus packet success rate (PSR), CCI versus SNR and CCI-NPLR. Due to the large fluctuating range of the raw CCI, Kalman filter is introduced to do de-noising of the raw CCI. The cubic model and the least squares method are employed to fit the relationship between CCI and SNR. In the experiments, many groups of comparison have been conducted and the results show that the proposed mechanism can achieve more accurate measurement of the non-perceived packet loss than existing approaches. Moreover, it has the advantage of decreasing extra energy consumption caused by sending large number of probe packets.