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This paper focuses on proposing approaches for the improvement of Voice over Internet Protocol (VoIP) service quality in Wireless Mesh Network (WMN). While WMNs have self-healing, self-forming and dynamic topology features, they still pose challenges for the implementation of multimedia applications, such as voice, in various scenarios. The research has been conducted using a network simulator and...
In this paper, we present experimental results on WebRTC voice quality as a function of LTE radio coverage. Different LTE radio conditions were tested by varying the radio path loss (with fast fading) in controlled test lab conditions. Voice quality was evaluated at different speech coding bit rates in terms of POLQA score and mouth-to-ear delay. Results show that, with acknowledged mode (AM) data...
The paper deals with the modelling of the network effects on the quality of speech in the Voice over IP networks. The main purpose of the ideas presented here is to achieve highprecision estimation of the speech quality in the environment in which the classical approaches of speech quality determination fail. To achieve such a high precision a modular neural network model is used to map the effects...
Voice over Internet Protocol (VoIP) has become a recent topic of research in both the Internet and the Telecommunication industry. The terrific increase in popularity of VoIP services is a result of huge growth in broadband usage. Both in wired as well as in wireless communication, VoIP is replacing the traditional telephony techniques. To ensure a good quality speech through VoIP, certain Quality...
Wireless Local Area Network (WLAN) has enabled greater communication capabilities compared to LAN counterpart. However, when it comes to quality of service (QoS), WLAN has lower reliability, where it has higher latency and packet-loss, especially for real-time streaming applications such as voice over internet protocol (VoIP) and multimedia. Thus, improving QoS in WLAN is a major research challenge,...
This paper presents the performance evaluation of Voice over Internet Protocol (VoIP) in Multi-hop Wireless Network (MWN) developed using Multi-radio Access Relay (MAR). The MWN is deployed using 3 MARs in Universiti Teknikal Malaysia campus. The performance of VoIP are investigated using Real Time Protocol (RTP) and Compress Real Time Protocol (CRTP) header techniques. RTP and CRTP are used to transport...
This paper investigates and compares the performance of the popular VoIP codecs G.711, G.723.1 and G.729A over an IP network based on SIP architecture, using RTP as a transport protocol. This is based on different simulations in order to evaluate the performance of each codec and to find the optimal codec. For this objective this paper analyzes Quality of service (QoS) parameters, principally delay,...
This paper proposes silence drop first algorithm(SDF) for the active buffer management. This algorithm finds and drops silence packet rather than talkspurt packet in the queue for resolving buffer overflow of queue. This algorithm can serve more simultaneous user while maintain voice service quality with same link capacity. Simulations with voice codec of G.711 and G.729a are performed in this paper...
Integrated Circuit(IC) design has seen a revolutionary progress in the past two decades with shrinking sizes of VLSI fabrication processes. This has an advantage of fabricating millions of transistors in a single chip IC. On the other hand it also creates many challenges in Deep Sub-Micron (DSM) technologies. One of the greatest challenges in DSM designs is inter-wire cross talk, which becomes significant...
There have recently been serious social issues involved in multimedia signal processing such as malicious attacks and tampering with digital audio/speech signals. Fragile speech watermarking is a technique that enables the detection of tampering with the original signals. We previously proposed an inaudible digital-audio watermarking approach based on cochlear delay. We investigated how the proposed...
This paper develops a simple analytical model based on a Markov chain that calculates saturation throughput and average packet delay for the IEEE802.11 Distributed Coordination Function (DCF) for network scenarios with voice and data stations. The analytical model is validated by comparing its results with simulation outcome. Based on the analytical model, a methodology that calculates voice capacity...
In the article we propose a method of improving ITU-T E-model MOS estimate of VoIP call quality. The improvement consists of including the effects of network jitter as measured and distributed in RTCP packets; jitter buffer size at receiver and codec packetization settings as input parameters of E-Model. Our method uses Pareto/D/1/k system for modeling general VoIP input traffic stream interarrival...
Screen is becoming a new dimension in cloud computing platforms, and low latency screen sharing in unreliable networks is becoming more and more important. Due to the different characteristics between the screen codecs and video codecs, current transmission technologies on low-latency video streaming cannot be directly applied to screen sharing. So in this paper we first theoretically analyze the...
Measuring and predicting users quality of experience (QoE) in dynamic network conditions is a challenging task. This paper presents results related to a decision-theoretic methodology incorporating Bayesian networks (BNs) and utility theory for quality of experience (QoE) measurement and prediction in mobile computing scenarios. In particular, we show how both generative and discriminative BNs can...
This work describes the factors limiting the performance in practical implementations of digital active noise cancellation (ANC) in commercial systems. This includes the codec delay, the secondary path variability, and the acoustic limitations. The causes for each limiting factor are studied and solutions are proposed that significantly improve system performance and robustness.
Many of the distributed video coding (DVC) systems described in the literature make use of a feedback channel from the decoder to the encoder to determine the rate. However, the number of requests through the feedback channel is often high, and as a result the overall delay of the system could be unacceptable in practical applications. As a solution, feedback-free DVC systems have been proposed, but...
Mobile voice-assisted services are currently experiencing strong growth. However, occasionally low real-time quality of service within mobile networks could have significant negative impact on quality of experience of users interacting with automated voice services. Latency may grow to unacceptable levels and speech recognition and synthesis might suffer. We present a methodology of mitigating such...
The article is analyzing the real impact audio and video streams for teleconferencing services in NGN networks. The test scenario is based on examining the data flow between Asterisk PBX and SIP client. This examination is performed by analysis of the data stream for the Asterisk PBX, depending on the parameters for audio / video stream to the SIP client. These parameters are represented using different...
Enhanced Distributed Channel Access (EDCA) is a mandatory part of the IEEE 802.11e standard which provides Medium Access Control (MAC) layer solution for Quality of Service (QoS) provisioning in Wireless Local Area Networks (WLANs). It is a contention based protocol which prioritizes channel access to QoS traffic using four Access Categories (ACs). In this paper, we introduce the performance evaluation...
Full audio bandwidth with very low algorithmic delay CODEC is state-of-the-art in CODEC technology. This type of CODEC is expected to support full frequency range for human hearing. It will enable future multipurpose audio applications, especially those which require high quality of audio signal with very low delay. A popular choice for this type of CODEC is the Constrained Energy Lapped Transform...
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