This paper investigates the impact of frequent and small playout delay adjustments (time-shifting) of 30 ms or less introduced to silence periods by Voice over IP (VoIP) jitter buffer strategies on listening quality perceived by the end... more
This paper investigates the impact of frequent and small playout delay adjustments (time-shifting) of 30 ms or less introduced to silence periods by Voice over IP (VoIP) jitter buffer strategies on listening quality perceived by the end user. In particular, the quality impact is assessed using both a subjective method (quality scores obtained from subjective listening test) and an objective method based on perceptual modelling. Two different objective methods are used, PESQ (Perceptual Evaluation of Speech Quality, ITU-T Recommendation P.862) and POLQA (Perceptual Objective Listening Quality Assessment, ITU-T Recommendation P.863). Moreover, the relative accuracy of both objective models is assessed by comparing their predictions with subjective assessments. The results show that the impact of the investigated playout delay adjustments on subjective listening quality scores is negligible. On the other hand, a significant impact is reported for objective listening quality scores predicted by the PESQ model i.e. the PESQ model fails to correctly predict quality scores for this kind of degradation. Finally, the POLQA model is shown to perform significantly better than PESQ. We conclude the paper by identifying further related research that arises from this study.
Um dos principais componentes da tecnologia de voz sobre IP são os algoritmos de codificação da fala, conhecidos como codecs. Existe uma grande variedade de codecs e alguns apresentam melhor qualidade que outros, a depender do grau de... more
Um dos principais componentes da tecnologia de voz sobre IP são os algoritmos de codificação da fala, conhecidos como codecs. Existe uma grande variedade de codecs e alguns apresentam melhor qualidade que outros, a depender do grau de perda de informação proporcionada pelas técnicas de compressão empregadas. Este trabalho tem por finalidade avaliar a qualidade de alguns codecs de código aberto, tendo como ponto de partida a implementação do procedimento de avaliação definido pela Rec.ITU-T P.834 para derivação do parâmetro de degradação da fala a 0% de perda de pacotes (Ie). Este parâmetro, específico para cada codec, é utilizado pelo Modelo E (Rec. ITU-T G.107) para avaliação da qualidade da fala.
ASTPP is an Open Source VoIP billing solution for Freeswitch. It supports pre-paid and post-paid billing with call rating, credit control and call reporting. It also provides many other features such as calling cards, least cost routing... more
ASTPP is an Open Source VoIP billing solution for Freeswitch. It supports pre-paid and post-paid billing with call rating, credit control and call reporting. It also provides many other features such as calling cards, least cost routing (LCR), did management, reseller management etc.
Header compression techniques can improve the speech quality of VoIP applications. However, its deployment over wireless mesh environments may not offer much improvement, since the wireless and multi-hop characteristics of such a network... more
Header compression techniques can improve the speech quality of VoIP applications. However, its deployment over wireless mesh environments may not offer much improvement, since the wireless and multi-hop characteristics of such a network impose a high packet loss rate. In this work, we propose a new approach for header compression, using packet aggregation to cooperate with the header compression scheme. Simulation results show that the new cooperative approach can decrease significantly the packet loss, impacting positively on the speech quality.
Monitoring speech quality in Voice over IP (VoIP) networks is important to ensure a minimal acceptable level of speech quality for IP calls running through a managed network. Information such as packet loss, codec type, jitter, end-to-end... more
