The impact of the non-uniform individual sensor node lifetime on the
connectivity of a data gathering tree over time is studied in this
research. The lifetime of sensor devices depends on the device failure rate and/or battery energy depletion, and surviving nodes may not preserve the uniform node density across the network as nodes age. We first examine the general node aging problem by considering the energy consumption rate and the node failure rate. The energy consumption rate in a data gathering tree is presented with or without data aggregation. The nodes in each hop level show a different energy depletion rates even with data aggregation, which is studied by mathematical analysis as well as simulation results. Then, the resulting non-uniform connectivity over time in a data gathering tree is examined with a node's survivor function. It is shown by mathematical analysis and simulation results that the node aging process has a significant impact on the connectivity
as the hop distance increases.
The recent ITU-T Recommendation P.862, known as the Perceptual
Evaluation of Speech Quality (PESQ) is an objective end-to-end
speech quality assessment method for telephone networks and speech
codecs through the measurement of received audio quality. To
ensure that certain network distortions will not affect the
estimated subjective measurement determined by PESQ, the algorithm
takes into account packet loss, short-term and long-term time
warping resulted from delay variation. However, PESQ does not work
well for time-scale audio modification or temporal clipping. We
investigated the factors that impact the perceived quality when
time-scale modification is involved. An objective measurement of
time-scale modification is proposed in this research, where the
cross-correlation values obtained from time-scale modification
synchronization are used to evaluate the quality of a time-scaled
audio sequence. This proposed objective measure has been verified
by a subjective test.
Traditional TCP performance degrades over lossy links, as the TCP sender assumes that packet loss is caused by congestion in the network path and thus reduces the sending rate by cutting the congestion window multiplicatively, and a mechanism to overcome this limitation is investigated in this research. Our scheme identifies the network path condition to differentiate whether congestion happens or not, and responds differently. The basic idea of separating congestion and non-congestion caused losses is to compare the estimated current available bandwidth and the average available bandwidth. To minimize the effect of temporary fluctuation of measurements, we estimate the available bandwidth with a higher weight on stable measurements and a lower weight on unstable fluctuations. In our scheme, packet loss due to congestion invokes the TCP Newreno procedure. In cases of random loss that is not related to congestion, the multiplicative decrease of the
sending rate is avoided to achieve higher throughput. In addition, each duplicate acknowledgement after a fast retransmission will increase the congestion window to fully recover its sending rate. Extensive simulation results show that our differentiation algorithm achieves high accuracy. Accordingly, the TCP connection over lossy link with the proposed scheme provides higher throughput than TCP Newreno.