Assessment of effects of packet loss on speech quality in VoIP

Ding, Lijing; Goubran, Rafik

doi:10.1109/have.2003.1244724

Cited by 41 publications

(21 citation statements)

References 3 publications

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“…It is known that perceived quality of packet loss concealed speech varies with codec and PLC types, packet size, loss pattern, and loss rate [7]. Our experiments suggest that PESQ and P.563 accuracy is also sensitive to such parameters.…”

Section: Plc Impairment Modelingmentioning

confidence: 62%

A Hybrid Signal-and-Link-Parametric Approach to Single-Ended Quality Measurement of Packetized Speech

Falk

Yuan

Chan

2007

2007 IEEE International Conference on Acoustics, Speech and Signal Processing - ICASSP '07

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A hybrid signal-and-link-parametric approach to single-ended quality measurement of packetized speech is proposed. Transmission link parameters are used to determine a base quality for the test signal. The base quality is adjusted by degradation factors calculated from perceptual features extracted from the test signal. The degradation factors are based on KullbackLeibler distances between a parametric model trained online for the extracted features and reference models of normative speech behavior. The proposed method overcomes the limitations of pure link parametric and pure signal-based methods.

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Section: Plc Impairment Modelingmentioning

confidence: 62%

A Hybrid Signal-and-Link-Parametric Approach to Single-Ended Quality Measurement of Packetized Speech

Falk

Yuan

Chan

2007

2007 IEEE International Conference on Acoustics, Speech and Signal Processing - ICASSP '07

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“…As the G.729 is a CELP-based codec, the use of linear predictive model of speech production can lead to variations in codec performance with different talkers or languages [18]. Support for such a speculation can be found in at least two studies using the same codec [6], [19], which, in case of zero packet loss and using different speech samples from the ITU-T data set [20], rendered measured I e values of 21.14 and 17.128, respectively, similar to the value obtained for this study. From the curves, a nonlinear regression model can be derived for each I e,k by the least-squares data fitting method.…”

Section: Multi-stream Voice Quality Prediction Modelmentioning

confidence: 99%

Perceptual-Based Playout Mechanisms for Multi-Stream Voice over IP Networks

Chang

Chiang

2011

IEICE Trans. Inf. & Syst.

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SUMMARYPacket loss and delay are the major network impairments for transporting real-time voice over IP networks. In the proposed system, multiple descriptions of the speech are used to take advantage of the packet path diversity. A new objective method is presented for predicting the perceived quality of multi-stream voice transmission. Also proposed is a multi-stream playout buffer algorithm, together with an adaptive parameter adjustment scheme, that maximizes the perceived speech quality via delay-loss trading. Experimental results showed that, compared to FECprotected single-path transmission, the proposed multi-stream transmission scheme achieves significant reductions in delay and packet loss rates as well as improved speech quality. key words: voice over IP, E-model, playout buffer

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“…On the other hand, E-Model has the following disadvantages: 1) its listening quality assessment is less accurate than that derived by a signal-based algorithm such as PESQ; 2) it does not consider the variability of network delays and loss rates; and 3) it does not take account of the interaction between different factors, e.g., the interplay between network delay and listening quality, or between network delay and loss rates. To address the above problems, Ding et al [5] proposed a hybrid model that integrates PESQ and E-Model and thereby provides the advantages of both models. We introduce this model in Section 5, and explain how we use it to evaluate the QoE of VoIP communications.…”

Section: Related Workmentioning

confidence: 99%

“…Therefore, we employ the QoE measurement model proposed by Ding et al [5] to quantify the overall QoE pro- …”

Section: Qoe Measurementmentioning

confidence: 99%

An empirical evaluation of VoIP playout buffer dimensioning in Skype, Google talk, and MSN Messenger

Chen

Huang

et al. 2009

Proceedings of the 18th International Workshop on Network and Operating Systems Support for Digital Audio and Video

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VoIP playout buffer dimensioning has long been a challenging optimization problem, as the buffer size must maintain a balance between conversational interactivity and speech quality. The conversational quality may be affected by a number of factors, some of which may change over time. Although a great deal of research effort has been expended in trying to solve the problem, how the research results are applied in practice is unclear.In this paper, we investigate the playout buffer dimensioning algorithms applied in three popular VoIP applications, namely, Skype, Google Talk, and MSN Messenger. We conduct experiments to assess how the applications adjust their playout buffer sizes. Using an objective QoE (Quality of Experience) metric, we show that Google Talk and MSN Messenger do not adjust their respective buffer sizes appropriately, while Skype does not adjust its buffer at all. In other words, they could provide better QoE to users by improving their buffer dimensioning algorithms. Moreover, none of the applications adapts its buffer size to the network loss rate, which should also be considered to ensure optimal QoE provisioning.

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Assessment of effects of packet loss on speech quality in VoIP

Cited by 41 publications

References 3 publications

A Hybrid Signal-and-Link-Parametric Approach to Single-Ended Quality Measurement of Packetized Speech

A Hybrid Signal-and-Link-Parametric Approach to Single-Ended Quality Measurement of Packetized Speech

Perceptual-Based Playout Mechanisms for Multi-Stream Voice over IP Networks

An empirical evaluation of VoIP playout buffer dimensioning in Skype, Google talk, and MSN Messenger

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