Sharif Digital Repository

In the internet telephony, loss of IP packets causes instantaneous discontinuities in the received speech. In this paper, we have focused on finding an error resilient method for this problem. Our proposed method creates artificial correlation between speech samples that pre-distorts the speech signal. The receiver uses this correlation to reconstruct the lost speech packets. An appropriate speech enhancement technique is designed for the reduction of the processing error in the recovered speech caused by the speech codecs. The SegSNR results show the superiority of our proposed speech enhancement method over a recently proposed one. © 2007 IEEE  

This paper introduces Niusha, the first Persian speech-enabled IVR platform. This platform uses Persian recognizer and Persian text-to-speech synthesizer engines in order to interact with users. The platform is designed in a way that it can simply be customized in various domains and its components are adjustable with new words  

The Session Initiation Protocol (SIP), which is becoming the de facto standard for the next-generation VoIP networks, is currently receiving much attention in many aspects. One aspect that was not deeply addressed in the original SIP is its authentication procedure. Apart from its security, an SIP authentication procedure should be efficient. This paper proposes a robust and efficient three-party SIP authentication protocol. In this protocol, the. end users are. authenticated with the proxy server in their domain using the. registrar server. Compared to previous works, our proposed protocol is more efficient and secure. To support our protocol with a formal security proof, its model is... 

Apart from its security, an SIP (Session Initiation Protocol) authentication protocol shall be efficient; because in order to replace traditional telephony, VoIP services have to offer enough security and QoS compared to PSTN services. Recently, Srinivasan et al. proposed an efficient SIP authentication scheme. The low delay overhead introduced by their scheme, causes the total call setup time to be well within the acceptable limit recommended by ITU-T. Based on their work, this paper proposes an SIP authentication scheme. In both schemes, the end users are authenticated with the proxy server in their domain using the registrar server. Comparing with the Srinivasan et al.'s scheme, our... 

In this paper, we introduce novel mechanisms for measuring and evaluating the performance of a network and its devices, configured especially for Voice over IP (VoIP). We study the effects of different configurations on final voice quality by enabling or changing the related equipment features. In our study we quantify quality changes based on different router configurations and discuss precisely to what extent and how much this quality is improved versus features or special configurations of network devices. These results can be used for two purposes: compare and benchmark different routers in order to know which one is proper for a typical VoIP network, and select the best configurations.... 

Quality of Experience (QoE) measures the overall quality of a service from users’ point of view by considering several system, human, and contextual factors. There exist various objective and subjective methods for QoE prediction. Although the subjective approach is more expensive and challenging than the objective approach, QoE’s level can be more accurately determined by a subjective test. This paper investigates various features affecting QoE by proposing a comprehensive subjective evaluation. First, we show that many unconsidered factors can significantly affect QoE. We have generated voice samples featuring different values for novel factors related to the speaker, signal, and network.... 

Speech enhancement in real-time applications improves the quality and intelligibility of the speech and reduces communication fatigue. Nowadays, due to reactivity of the systems and spread of online real-time applications, including VoIP, state-space models have been used broadly. This paper presents a speech enhancement method based on adaptive Bayesian-Kalman filter and Bayesian-MAP estimation to improve the performance and the quality of the enhancement procedure. The enhancement method includes a combination of Bayesian-Kalman filter for noise reduction and Bayesian-MAP estimation for parameter estimation of the lost speech segments. Performance evaluation and result of the proposed... 

A major problem in real-time packet-based communication systems, is misrouted or delayed packets which results in degraded perceived voice quality. If packets are not available on time, the packets are considered as lost. The easiest solution in a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system, to avoid degradation in speech quality due to packet loss, a suitable method or algorithm is needed to replace the missing segments of speech. In this paper, we introduce an adaptive filter for replacement of lost speech segment. In this method Kalman filter as a state-space based method will be used to predict the... 

A major problem in real-time packet-based communication systems, is misrouted or delayed packet which results in degraded perceived voice quality. If packets are not available on time, the packets are considered lost. The easiest solution in a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system, to avoid degradation in speech quality due to packet loss, a suitable method or algorithm is needed to replace the missing segments of speech. In this paper, we introduce an adaptive filter for replacement of lost speech segment. In this method Kalman filter as a state-space based method will be used to predict the clean... 

A major problem in real-time packet-based communication systems, is misrouted or delayed packets which results in degraded perceived voice quality. If packets are not available on time, the packets are considered as lost. The easiest solution in a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system, to avoid degradation in speech quality due to packet loss, a suitable method or algorithm is needed to replace the missing segments of speech. In this paper, we introduce an adaptive filter for replacement of lost speech segment. In this method Kalman filter as a state-space based method will be used to predict the clean... 

In this paper, a low order recursive linear prediction method and recursive least square as an adaptive filter (LP-RLS) are introduced to predict the speech and the excitation signals. In real-time packet-based communication systems, one major problem is misrouted or delayed packets which results in degraded perceived voice quality. If packets are not available on time, the packet is known as lost packet. The easiest task of a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system, to avoid quality reduction due to packet loss, a suitable method and/or algorithm is needed to replace the missing segments of speech. The... 

In real-time packet-based communication systems, one major problem is misrouted or delayed packets which results in degraded perceived voice quality. If packets are not available on time, the packet is known as lost packet. The easiest task of a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system, to avoid quality reduction due to packet loss, a suitable method and/or algorithm is needed to replace the missing segments of speech. In this paper, we introduce a low order recursive linear prediction method for replacement of lost speech segment. In this method a normalized least mean square (NLMS) as an adaptive filter... 

VoIP over WLAN is a promising technology as a powerful replacement for current local wireless telephony systems. Packet timing Jitter is a constant issue in QoS of IEEE802.11 networks and exploiting an optimum jitter handling algorithm is an essential part of any VoIP over WLAN (VoWiFi) devices especially for the low cost devices with limited resources. In this paper two common algorithms using buffer as a method for Jitter handling are analyzed with relation to different traffic patterns. The effect of different buffer sizes on the quality of voice will be assessed for these patterns. Various traffic patterns were generated using OPNET and Quality of output voice was evaluated based on ITU...