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Recently user quality of experience (QoE) is employed in evaluating end user satisfaction in communications systems. Generally, current approaches for QoE assessment are obtrusive, laboratory based and offline. Estimation of user satisfaction in static manner based on mean opinion score is not directly related to instantaneous individual end user contentment. In this paper, based on correlations between user's physiological signals and her/his feelings about the service quality, a non-intrusive and user centric QoE assessment system for voice communications is developed. The findings of this study indicate that the emotional patterns in response to the changes in channel quality can be adapted to estimate the level of satisfaction in a QoE assessment system in a live manner. Based on experimental results, two categories of users are identified: sensitive and insensitive towards quality degradations. The results indicate that for the sensitive users, our non-intrusive subjective quality assessment method outperforms ITU-T P.563 standard with respect to root mean square error; while, the results are much better among the insensitive users. [ABSTRACT FROM AUTHOR]

The network topology involves a series of conditions: the type of the cable used, the route of the cable, the elements of virtualisation, the connections of the network assets, the layout of the encryption elements. The network topology can determine the way the computers in the network communicate. In the domain of local networks, in the author's opinion, there are several possible topologies, out of which only three are widespread: line, ring and star. The mission networks should facilitate the access to the services provided by the infrastructure and ISTAR (Intelligence, Surveillance, Target Acquisition and Reconnaissance) in the deployment area. [ABSTRACT FROM AUTHOR]

A key feature of wireless mesh networks is that multiple independent paths through the network are available. Multiple descriptions coding is often suggested as a source coding scheme to take advantage of this path diversity. We compare multiple description (MD) coding with path diversity (PD) against a full-rate single description (SD) coder without PD, and two simple PD methods of 1) repeating a half-rate SD coder over both paths and 2) repeating the full-rate parent SD coder over the two paths. We first present a theoretical analysis comparing the average distortion per symbol in packetized communication using the above mentioned MD and PD methods to transmit a memoryless Gaussian source over additive white Gaussian noise channels. Next, using two new MD speech coders with balanced side descriptions derived from the AMR-WB and G.729 standards, we evaluate delivered voice quality using PESQ-MOS and compare MD coding against the PD methods for random and bursty packet losses. Both the theoretical analyses and the speech coding experiments show that with packet overheads, the simple PD methods may be preferable to MD coding. A new performance measure that incorporates both quality and bit rate is shown to account for the tradeoffs more explicitly. [ABSTRACT FROM PUBLISHER]

The quality of service limitation of today's Internet is a major challenge for real-time voice communications. Excessive delay, packet loss, and high delay jitter all impair the communication quality. A new receiver-based playout scheduling scheme is proposed to improve the tradeoff between buffering delay and late loss for real-time voice communication over IP networks. In this scheme the network delay is estimated from past statistics and the playout time of the voice packets is adaptively adjusted. In contrast to previous work, the adjustment is not only performed between talkspurts, but also within talkspurts in a highly dynamic way. Proper reconstruction of continuous playout speech is achieved by scaling individual voice packets using a time-scale modification technique based on the Waveform Similarity Overlap-Add (WSOLA) algorithm. Results of subjective listening tests show that this operation does not impair audio quality, since the adaptation process requires infrequent scaling of the voice packets and low playout jitter is perceptually tolerable. The same time-scale modification technique is also used to conceal packet loss at very low delay, i.e., one packet time. Simulation results based on Internet measurements show that the tradeoff between buffering delay and late loss can be improved significantly. The overall audio quality is investigated based on subjective listening tests, showing typical gains of 1 on a 5-point scale of the Mean Opinion Score. [ABSTRACT FROM PUBLISHER]