International audienceThis survey gives a comprehensive review of recent advances related to the topic of VoIP QoE (Quality of user' Experience). It starts by providing some insight into the QoE arena and outlines the principal building blocks of a VoIP application. The sources of impairments over data IP networks are identified and distinguished from signal-oriented sources of quality degradation observed over telecom networks. An overview of existing subjective and objective methodologies for the assessment of the QoE of voice conversations is then presented outlining how subjective and objective speech quality methodologies have evolved to consider time-varying QoS transport networks. A description of practical procedures for measuring VoIP QoE and illustrative results is then given. Utilization methodology of several speech quality assessment frameworks is summarized. A survey of emerging single-ended parametric-model speech quality assessment algorithms dedicated to VoIP service is then given. In particular, after presenting a primitive single-ended parametric-model algorithm especially conceived for the evaluation of VoIP conversations, new artificial assessors of VoIP service are detailed. In particular, we describe speech quality assessment algorithms that consider, among others, packet loss burstiness, unequal importance of speech wave, and transient loss of connectivity. The following section concentrates on the integration of VoIP service over mobile data networks. The impact of quality-affecting phenomena, such as handovers and CODEC changeover are enumerated and some primary subjective results are summarized. The survey concludes with a review of open issues relating to automatic assessment of VoIP

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A system based on Raman spectroscopy has been developed for closed-loop control during the manufacture of titanium dioxide (TiO2). Spectra were obtained in the 50 to 700 cm−1 range using excitation at ∼532 nm and a fixed transmission-grating spectrograph coupled to a CCD detector. Quantitation has been achieved by linear interpolation based on the integrated intensity of phonons at 142 and 610 cm−1 due to anatase and rutile phases, respectively. The spectroscopic instrumentation has been integrated with a powder sampling system and installed on four full-scale manufacturing plants. Fiber coupling (up to 100 m) has been employed, and each spectrometer system is multiplexed to three sampling heads. Spectra are corrected for changes in sample temperature, and the systems are designed to be fully automatic and require no user intervention.

Delivering PSTN-like quality over current besteffort Internet infrastructure presents many technical challenges. Much research in recent years has focused on receiver-based approaches which adapt to varying network conditions in order to optimize playout quality. In this paper, we propose and evaluate a receiver-based approach that implements a hybrid adaptivefixed playout regime by integrating synchronized time into the playout algorithm. Such an approach can deliver significantly better quality than existing adaptive techniques particularly when the underlying network is not heavily congested and end-toend delays are not excessive. We present some initial results from our testbed system using the ITU-T E-model to quantify improvements.

Receiver-based strategies focus on optimizing the buffer design (and thus application delay) to strike the correct balance between packet loss due to late arrival and overall delay.A larger receiver buffer means that the receiver can wait longer for delayed packets, but it also increases the overall end-toend delay. Setting a fixed buffer size in a besteffort environment is unsatisfactory because of its inflexibility, so a lot of research focuses on adaptive buffer strategies.