Packetized-voice/data integrated transmission on a token passing ring local area network

This paper investigates the performance of a token passing ring network with packetized-voice/data mixed traffic through extensive simulations. Both data and voice users are modeled in the simulations. Data users produce bursty traffic. Voice traffic is modeled as having alternating talkspurts and silences, with generation of voice packets at a constant rate during talkspurts and no packet generation during silence periods.The network performance measures obtained include: the distribution of transmission delays for voice packets, the average transmission delay and loss probabilities for voice packets, the number of voice users allowed on a network while satisfying the real-time constraints of speech, and the average transmission delay for data packets.Token passing ring local area networks are shown to effectively handle both voice and data traffic. The effects of system parameters (e.g., voice packet length, talkspurt/silence lengths, data traffic intensity, and limited versus exhaustive service disciplines) on network performance are discussed

Packet-switched voice and its application to integrated voice / data networks

Imperial Users onl

Development and evaluation of packet video schemes

Reflecting the two tasks proposed for the current year, namely a feasibility study of simulating the NASA network, and a study of progressive transmission schemes, are presented. The view of the NASA network, gleaned from the various technical reports made available to use, is provided. Also included is a brief overview of how the current simulator could be modified to accomplish the goal of simulating the NASA network. As the material in this section would be the basis for the actual simulation, it is important to make sure that it is an accurate reflection of the requirements on the simulator. Brief descriptions of the set of progressive transmission algorithms selected for the study are contained. The results available in the literature were obtained under a variety of different assumptions, not all of which are stated. As such, the only way to compare the efficiency and the implementational complexity of the various algorithms is to simulate them

A Hybrid Signal-and-Link-Parametric Approach to Single-Ended Quality Measurement of Packetized Speech

A hybrid signal-and-link-parametric approach to single-ended quality measurement of packetized speech is proposed. Trans-mission link parameters are used to determine a base quality for the test signal. The base quality is adjusted by degradation factors calculated from perceptual features extracted from the test signal. The degradation factors are based on Kullback-Leibler distances between a parametric model trained online for the extracted features and reference models of normative speech behavior. The proposed method overcomes the limita-tions of pure link parametric and pure signal-based methods. Index Terms — Quality measurement, VoIP, packet loss concealment, Kullback-Leibler distance

A Statistical Approach to Adaptive Playout Scheduling in Voice Over Internet Protocol Communication

Factors like network delay, latency and bandwidth significantly affect the quality of communication using Voice over Internet Protocol. The use of jitter buffer at the receiving end compensates the effect of varying network delay up to some extent. But the extra buffer delay given for each packet plays a major role in playing late packets and thereby improving voice quality. As the buffer delay increases packet loss rate decreases, which in general is a very good sign. However, an increase of buffer delay beyond a certain limit affects the interactive quality of voice communication. In this paper, we propose a statistical framework for adaptive playout scheduling of voice packets based on network statistics, packet loss rate and availability of packets in the buffer. Experimental results show that the proposed model allocates optimal buffer delay with the lowest packet loss rate when compared with other algorithms

Practical Distributed Video Coding in Packet Lossy Channels

Improving error resilience of video communications over packet lossy channels is an important and tough task. We present a framework to optimize the quality of video communications based on distributed video coding (DVC) in practical packet lossy network scenarios. The peculiar characteristics of DVC indeed require a number of adaptations to take full advantage of its intrinsic robustness when dealing with data losses of typical real packet networks. This work proposes a new packetization scheme, an investigation of the best error-correcting codes to use in a noisy environment, a practical rate-allocation mechanism, which minimizes decoder feedback, and an improved side-information generation and reconstruction function. Performance comparisons are presented with respect to a conventional packet video communication using H.264/advanced video coding (AVC). Although currently the H.264/AVC rate-distortion performance in case of no loss is better than state-of-the-art DVC schemes, under practical packet lossy conditions, the proposed techniques provide better performance with respect to an H.264/AVC-based system, especially at high packet loss rates. Thus the error resilience of the proposed DVC scheme is superior to the one provided by H.264/AVC, especially in the case of transmission over packet lossy networks

Description and performance of a digital mobile satellite terminal

A major goal of the Mobile Satellite Experiment (MSAT-X) program at the Jet Propulsion Lab (JPL) is the development of an advanced digital terminal for use in land mobile satellite communication. The terminal has been developed to minimize the risk of applying advanced technologies to future commercial mobile satellite systems (MSS). Testing with existing L band satellites was performed in fixed, land mobile and aeronautical mobile environments. JPL's development and tests of its mobile terminal have demonstrated the viability of narrowband digital voice communications in a land mobile environment through geostationary satellites. This paper provides a consolidated description of the terminal architecture and the performance of its individual elements

ROBUST DECODING OF A 3D-ESCOT BITSTREAM TRANSMITTED OVER A NOISY CHANNEL

International audienceIn this paper, we propose a joint source-channel (JSC) decoding scheme for 3D ESCOT-based video coders, such as Vidwav. The embedded bitstream generated by such coders is very sensitive to transmission errors unavoidable on wireless channels. The proposed JSC decoder employs the residual redundancy left in the bitstream by the source coder combined with bit reliability information provided by the channel or channel decoder to correct transmission errors. When considering an AWGN channel, the performance gains are in average 4 dB in terms of PSNR of the reconstructed frames, and 0.7 dB in terms of channel SNR. When considering individual frames, the obtained gain is up to 15 dB in PSNR