This article investigates prediction of Quality of Experience (QoE) by comparing borrowing-from-neighbor situations and isolated ones. We demonstrate that joint utilization of multiple QoE measures enhances the accuracy of QoE prediction compared to that by a collection of individual QoE measures each regressed on Quality of Service (QoS) parameters, while the accuracy improvement with additional usage of QoS information in the former is limited. As an example of system that needs multidimensional QoE representation, the article gives haptic audiovisual interactive communications. We employ QoE and QoS data taken previously in an experiment, where 13 QoE measures (a five-point score each) and 12 QoS parameters (nonnegative continuous values each) are available at three average rates of load traffic. We build two kinds of Bayesian models for QoE prediction; one is a logistic regression model of a single QoE measure as the response variable and QoS parameters as predictors, which is a typical traditional method of discrete QoE prediction and isolated in a sense. The other is a structural equation model (SEM) with latent constructs (i.e., factors) of audiovisual quality, haptic quality, and user experience quality; the original SEM, which contains only QoE indicators of three constructs (audiovisual quality (AVQ), haptic quality (HQ), and user experience quality (UXQ)), was proposed in one of the author’s previous studies. This article extends the SEM to accommodate QoS parameters. We develop two kinds of new SEMs with QoS parameters: One has three extended constructs referred to as eAVQ, eHQ, and eUXQ, each of which has both QoE and QoS indicators, and the other has separate constructs for QoE and QoS, which lead to totally six constructs (AVqoe, Hqoe, UXqoe, AVqos, Hqos, and UXqos). We performed Markov chain Monte Carlo simulation of the Bayesian models with the JAGS software in an R environment. For comparison of QoE prediction accuracy, we adopt the 10-fold cross validation method and in part widely applicable information criterion. We then found that the three-construct models outperform the logistic regression models with respect to all subjective QoE measures and that the two kinds of the models are comparable as for the objective measure. The six-construct model exhibits almost the same accuracy as that of the three-construct one unless the number of QoE measures ( n qoe ) in the model is small. When the number n qoe is small, single-construct models may be a better choice. We have thus learned that multiple QoE measures should be utilized jointly (i.e., borrowing from neighbor) in QoE prediction rather than resorting to QoS information only.

This article proposes an empirical method for inferring causal directions in multidimensional Quality of Experience (QoE) in multimedia communications, noting that causation in QoE is perceptual. As an example for modeling framework, we pick up a Bayesian structural equation model (SEM) previously built for haptic audiovisual interactive communications. The SEM includes three constructs (Audiovisual quality, Haptic quality, and User experience quality), which are latent variables each representing a group of observed variables with similar characteristics. In the SEM, the causal directions of the constructs were assumed by resorting to the domain knowledge. This article aims at proposing a methodology for inferring causal directions of constructs in general by verifying the assumption of causal directions in the SEM through their observed data alone. For that purpose, we compare six SEMs each with different causal directions of constructs, one of which is the one from the domain knowledge. The proposed method is based on QoE prediction by a Bayesian approach with Markov chain Monte Carlo (MCMC) simulation. Setting observed scores to the indicators of exogenous variables in each SEM, we predict values of all the indicators; we then assess the mean square error (MSE) between predicted QoE and mean opinion score (MOS) from observed scores and estimate the probability distribution of the MSE in each SEM. We can compare any two SEMs to find which is more plausible by examining the probability that the MSE for one SEM is smaller than or equal to that for the other. These probabilities are estimated with MCMC simulation. The method indicates that the causal directions thus inferred for the haptic audiovisual interactive communications adequately support the original ones drawn from the domain knowledge. In addition, we demonstrate that QoE can behave like the “impact-perceive-adapt” model of the effects of delayed haptic and visual feedback on performance in a collaborative environment, which Jay, Glencross, and Hubbold proposed in 2007, and that it accompanies reversal of plausible causal directions like a flip–flop.

This article proposes a methodology for building and verifying plausible models that can express causation in multidimensional QoE for haptic-audiovisual interactive communications. For the modeling, we utilize subjective experimental data of five-point scores collected in a previous study where a pair of subjects carry out two kinds of interactive tasks (castanets hitting and object movement) in real space (not in virtual space). The multidimensional QoE is composed of 15 measures for the castanets hitting and 14 measures for the object movement. To reduce the dimension, we classify the QoE measures into three groups as indicators of three constructs (latent variables or factors): AVQ (AudioVisual Quality), HQ (Haptic Quality), and UXQ (User eXperience Quality). We then build two models: (1) a structural equation model in which AVQ and HQ correlated with each other give causal effects on UXQ, and (2) a confirmatory factor analysis model in which the three constructs are only correlated with each other. We refer to the former as 3C-SEM and the latter as 3C-CFA. We further introduce a CFA model with a single construct for which all QoE measures are its indicators (1C-CFA). We perform Bayesian analysis of the three models by means of Markov chain Monte Carlo simulation; in each model, the deviance information criterion is obtained for model comparison, and the posterior predictive p -value is calculated for model checking. As a result, we find that 3C-SEM is the most plausible and that HQ has a stronger causal effect on UXQ than AVQ. We also learn that the correlation between AVQ and UXQ is much higher than the direct causal effect and that the increase in the association as correlation is due to the causal effect of HQ on UXQ through the correlation of AVQ with HQ. Thus, it is suggested that improving haptic performance is more effective in enhancement of QoE than improving audiovisual performance.

This paper proposes a methodology for Bayesian modeling of multidimensional ordinal categorical data for QoE by means of the structural equation modeling (SEM) technique, focusing on the identifiability (indeterminacies) problem of latent variables. As an example material, we adopt a haptic-audiovisual interactive communication system along with previously collected data of 13 subjective QoE measures and a single objective QoE measure. To reduce the dimensionality of the model, we introduce two exogenous latent constructs, Audiovisual quality (AVQ) and Haptic quality (HQ), and an endogenous construct User experience quality (UXQ). For modeling of the categorical data of scores, we also employ continuous latent variables underlying the scores. We resort to the unit loading (UL) method and the unit variance (UV) method for the resolution of scale indeterminacies. Bayesian estimates of free parameters including the loading and regression coefficients by the two methods have different values, while estimates of the QoE measures have almost the same values, which are found to be satisfactory fit in terms of the overall posterior predictive p-values. A comparison of the two methods in terms of the p-value and DIC demonstrates that the model with UL constraints is somewhat better than the model with UV. We also learn that HQ has much stronger effects on UXQ than AVQ regardless of the identifiability constraint (UL or UV).

This paper presents a trial of establishing a framework for Bayesian statistical modeling of quality of experience (QoE) estimation and prediction in multimedia IP communications. As an example, we take a bandwidth guaranteed interactive audiovisual communication system with a QoE enhancement mechanism. By Bayesian statistical analysis, we demonstrate how QoE is affected by factors of the channel bandwidth, contents of tasks, customization of playout buffering control for QoE enhancement, users' individualities, and gender. Adopting a latent variable approach, we build Bayesian regression models with covariates of the factors and random effect terms having hierarchical priors for the users' attributes. We estimate the posterior probabilities of parameters in the models by performing Markov chain Monte Carlo simulations with WinBUGS. From among the models, we select the most plausible one and explore the sensitivity to priors. We then evaluate the QoE measure, which is the posterior mean of overall satisfaction measured as five-point scores, and examine the effects of the factors on the QoE through odds ratios. Also, the Bayesian model makes predictions of score frequencies and compares them with measured ones to reveal high accuracy of the prediction. In addition, a cross-validatory approach to the model checking is taken for investigating the adequacy of the model; the posterior predictive p-values of a discrepancy function of the model indicate that the model is satisfactory.

This paper proposes a methodology for assessing multidimensional QoE (Quality of Experience) by Bayesian analysis of Structural Equation Models (SEMs). As an example of the assessment, we employ a haptic-audiovisual interactive communication system because of its QoE's multidimensionality due to different traffic characteristics between the haptic media and audio-video streams. We deal with 14 dimensional QoE measures which consist of 13 subjective measures on a five-point scale and a single objective measure of efficiency; they are correlated with each other. In order to simplify the multidimensional causal relationships, we classify the QoE measures into three groups corresponding to three constructs (latent variables): AVQ (AudioVisual Quality), HQ (Haptic Quality) and UXQ (User experience Quality). We build a Bayesian SEM with the three latent variables and the 14 observed variables (QoE measures). We infer posterior probabilities of parameters in the model by Markov chain Monte Carlo (MCMC) simulation. As a result, we find that UXQ, which includes the QoE measure of overall satisfaction as an observed variable, is dominated by HQ and that the effect of AVQ on UXQ is small. We also notice that although the objective QoE (efficiency) is correlated with subjected QoE, a separate measure is necessary.

This paper presents a Bayesian statistical approach to QoE (Quality of Experience) assessment problems in multimedia communications. In order to explain how to apply the Bayesian method to the problem, we pick up, as an example, an interactive audiovisual communication with a QoE enhancement mechanism, which we call user-assisted QoE enhancement. Utilizing experimental data measured on subjects (users) in the interactive communication system, we first set up a framework of the QoE assessment in the context of Bayesian modeling.We build a couple of Bayesian hierarchical models with two-levels, user and gender, based on multivariate ordinal categorical logistic regression using a latent variable. We then perform Markov Chain Monte Carlo (MCMC) simulations with WinBUGS and check the convergence to obtain posterior probabilities of model parameters. Utilizing the posterior probabilities, we calculate QoE measures with 95% credible intervals and compare them with the traditional MOS (Mean Opinion Score) with 95% confidence intervals. We also examine the effects of video bandwidth, the kinds of tasks, the user-assisted QoE enhancement mechanism on QoE by taking into consideration the subjects' gender or equivalently his/her technical background of computer in this case.

This paper deals with audiovisual and haptic interactive IP communications and designs a multidimensional QoE monitoring system with the QoE estimation in real time. For the estimation, we carry out regression analysis between QoE and application-level QoS and obtain equations for the estimation. We measured the application-level QoS and made a subjective experiment by assessing QoE multidimensionally with the SD method. To monitor QoE, we derive estimates for three QoE metrics. We apply linear and non-linear regression analysis to the experimental results. As a result, we find that the obtained equations can estimate QoE with high accuracy. By using the equations, we have built the QoE monitoring system.

In this paper, we study a real-time estimation method of QoE of Multi-View Video and Audio (MVV-A) IP transmission. The method utilizes application-level QoS parameters of video which can be measured in real time. At first, we perform an experiment with various types of average load, playout buffering time, and additional delay to the network. In the experiment, we employ two contents and two user interfaces for viewpoint change. We assess QoE multidimensionally and then carry out linear regression analysis in order to obtain regression lines for estimating QoE. We use the QoE metrics as the dependent variables and employ the application-level QoS parameters as the independent variables. From the comparison of measured values and estimated ones, we notice that real-time estimation of QoE is feasible in MVV-A IP transmission.

This paper studies a method of enhancing QoE (Quality of Experience) in H.264-video and audio IP transmission by jointly applying FMO (Flexible Macroblock Ordering) and SCS (Switching between error Concealment and frame Skipping), which is a QoE-based video output scheme simply based on the receiver's operation. We conducted objective and subjective experiment on the method for combinations of two slice group map types (interleaved and dispersed), four contents and three picture patterns (I, I+4P's and I+14P's); we measured QoS (Quality of Service) and then assessed QoE in terms of the psychological scale. As a result, we find that for the patterns I+4P's and I+14P's, combinations of SCS and the dispersed map type work best, whereas for the pattern I, the frame skipping only (i.e., SCS without FMO) achieves the highest QoE.

This paper makes multidimensional QoE comparisons of GUIs for threshold selection in the QoE-based video output scheme SCS in audio-video IP transmission. SCS switches between error concealment and frame skipping by comparing the ratio of video slice loss in a video frame with a threshold value. For the purpose of adapting SCS to individual users' inclination, we need to create some appropriate threshold selection interfaces and give the option to select a threshold value to the user. We propose two new interfaces: a slide bar method (choosing a threshold value by a slide bar), and a two mode method (selection of one out of two modes by two buttons). In the latter, a threshold value is set according to the mode. To assess QoE multidimensionally, we conducted subjective experiments on four methods: the conventional error concealment method (100% method), a previously proposed interface called the radio button method (selecting a threshold value by four radio buttons) and the two new interfaces. As a result, we observe that the two new interfaces can achieve higher overall audiovisual QoE than the conventional 100% method and the radio button method. In addition, we show that the slide bar method achieves the highest controllability whereas it imposes a burden on the user and that the two mode method imposes the lightest burden on the user among the three interfaces while it provides high QoE.

This paper studies QoE (Quality of Experience) enhancement of audio-video IP transmission over the uplink channel, i.e., from subscriber stations (SSs) to the base station (BS), in the IEEE 802.16 BE service. We assume two types of capacity allocation schemes for uplink and downlink burst durations: static and adaptive. Furthermore, we introduce a piggyback request mechanism for uplink bandwidth requests from SSs to the BS in addition to a random access-based request mechanism. We assess QoE of audio-video streams for four schemes obtained from the combination of the bandwidth request mechanisms and the capacity allocation schemes. We also employ two types of audio-video contents. From the assessment result, we notice that the piggyback request mechanism can enhance QoE of audio-video transmission. In addition, the adaptive allocation scheme is effective for QoE enhancement particularly under heavily loaded conditions because of its efficient usage of OFDM symbols. We also find that the effects of piggyback request mechanism and capacity allocation schemes on QoE change according to the content types.

SCS is a video output scheme which is designed to maximize QoE in audio-video IP transmission; it is featured by the applicability to end-terminals independent of the type of used networks. SCS switches between video error concealment and video frame skipping by comparing the percentage of error-concealed video slices in a frame to a threshold value, which is a key parameter for successful implementation of SCS. This paper proposes two practical methods of threshold selection for SCS: a table lookup method and a user selection method. We compare the two proposed methods with the pure error concealment and the pure video frame skipping. We conducted subjective experiments on the four methods for combinations of several contents and picture patterns and then evaluated QoE in terms of the psychological scale. As a result, we observe that the table lookup method and the user selection method can achieve higher QoE than the other two conventional methods.

This paper proposes a method of customizing interactive audiovisual communication services over bandwidth guaranteed IP networks typified by NGN in order to maximize QoE(Quality of Experience) on an individual users' basis. The method is based upon an extension of a QoE guarantee architecture GPSQ (Guarantee of Psychologically Scaled Quality), which the authors proposed previously. GPSQ extended for interactive services is referred to as interactive GPSQ, which focuses on a salient feature associated with the interactive services, i.e., the existence of the optimum playout buffering time for absorbing delay jitter. The customization method is “semi-tailored” in the sense that it provides the users with a means of adjusting the buffering time so that it can maximize his/her own QoE. This paper examines the feasibility of the basic idea for interactive GPSQ and the customization method through a minimum number of conversational opinion tests, i.e., two tasks with different visual effects. For that purpose, we have built a simple experimental system of interactive GPSQ and made experiment on QoE assessment with and without the customization method. The experiment showed that the interactive GPSQ is feasible and that the customization improves QoE.

This paper identifies factors affecting QoE (quality of experience) of interactive audiovisual communications over bandwidth guaranteed IP networks and finds how to maximize QoE in that environment. Among the factors, we focus on the guaranteed bandwidth between the terminals, video encoding bit rate, the playout buffering time, and task. Under the condition that the guaranteed bandwidth for video is kept constant for a given amount of the guaranteed bandwidth for audio, we made experiment on QoE assessment for various values of the video encoding bit rate and playout buffering time. We suppose two kinds of task: one is audio-dominant, while the other is a task whose video affects QoE more than the former. As a result of the experiment, we see that an appropriate selection of the pair of the video encoding bit rate and playout buffering time maximizes QoE for the guaranteed bandwidth for video. Furthermore, we notice that the playout buffering time should be carefully chosen especially for the latter task.

This paper proposes a packet scheduling scheme for QoS support in audio-video transmission with IEEE 802.11e HCCA. In the proposed scheme, the access point (AP) basically allocates transmission opportunity (TXOP) for each station like the reference scheduler of the IEEE 802.11e standard, which is referred to as the TGe scheme in this paper. However, the AP with the proposed scheme can allocate additional TXOP in a service interval (SI) if the AP has received corrupted MAC protocol data units (MPDUs) in the previous SI. In addition, the proposed scheme calculates the number of MAC service data units (MSDUs) arrived in an SI, considering the inter-arrival time of audio samples and that of video frames, which are referred to as media units (MUs), at the application-layer. We then compare the proposed scheme with the TGe scheme in terms of application-level QoS in the case where stations transmit audio and video flows to the AP in a noisy environment. Numerical results show that the proposed scheme improves the average MU delay and media synchronization quality even if channel transmission error occurs to some extent. Since the proposed scheme uses the surplus bandwidth for MPDU retransmission, we also discuss the number of admitted stations in the scheme.

This paper compares QoE (Quality of Experience) of audio-video IP transmission with IEEE 802.11e EDCA and that with DCF in mobile ad hoc networks. We also assess QoE of the transmission with TXOP-bursting, which is an optional function of EDCA. QoE assessment of audio-video IP transmission in mobile ad hoc networks involves difficult issues. The wireless nodes can move, and then channel condition changes dynamically. The routing paths may also change as the wireless nodes move. In this paper, we tackle the difficulty of the assessment by means of a continuous time QoE assessment method, which utilizes the method of successive categories continuously in time. For the assessment, at first, we performed computer simulation, where a mesh topology network with mobile nodes is assumed for transmitting audio, video, and data traffic. We then carried out subjective experiment. The assessment results show that EDCA and TXOP enhance QoE of audio-video transmission over mobile ad hoc networks.

This paper proposes a methodology of video-stream output at the receiver for enhancing QoE (quality of experience) in audio-video IP transmission. The methodology copes with video packet loss by error concealment and/or video frame skipping; it utilizes the tradeoff relation of QoE between spatial and temporal quality caused by the two techniques. As a simple example of the methodology, we adopt a scheme which switches between the two techniques according to the percentage of video slices error-concealed in a frame; the scheme is referred to as SCS (switching between error concealment and frame skipping). We conducted experiments on the SCS with six contents. Taking into consideration the cross-modal interaction between audio and video, we then assessed QoE in terms of the psychological scale, which is more accurate than MOS (mean opinion score). The experimental result shows that the SCS can improve QoE over the simple error concealment or frame skipping by selecting an appropriate threshold value of the error concealment ratio, which depends on the content type, video picture pattern and degree of video motion.

This paper shows the effectiveness and feasibility of a video output scheme the authors proposed previously to enhance QoE (Quality of Experience) (i.e., perceptual QoS) of audio-video IP transmission. The scheme, SCS (Switching between error Concealment and frame Skipping), utilizes a tradeoff relation between spatial and temporal quality caused by video error concealment and video frame skipping, both of which cope with video packet loss. The scheme switches from error concealment to frame skipping when the percentage of video slices error-concealed in a frame exceeds a threshold value, which is a key to achieving high QoE with SCS. We propose a way of selecting appropriate threshold values on the basis of real-time estimation of QoE. For that purpose, we performed two experiments for four values of the threshold. One experiment is the measurement of application-level QoS and QoE, from which we confirm the effectiveness of SCS and derive multiple regression lines that estimate QoE from the application-level QoS. The other is the measurement of the percentage of the selected threshold value by the proposed way with the multiple regression lines. Examining the QoE measured by the first experiment when we adopt the selected threshold values in the second experiment, we find that the way gives appropriate selections in many cases and therefore the SCS with the way of threshold selection is feasible.

This paper estimates quality of experience (QoE), i.e., user-level QoS from MAC-level QoS for audio-video transmission over an IEEE 802.11e EDCA wireless LAN. The MAC-level QoS in this case has already been studied extensively; however, it may not provide exact information on QoE, which we can obtain by the estimation. We first examine MAC-level QoS by simulation in the case where wireless stations transmit audio and video flows to an access point (AP) in the presence of transmission errors. We then assess QoE by the method of successive categories, which is a psychometric method, and perform QoS mapping between MAC-level and user-level with multiple regression analysis. As a result, we obtain multiple regression lines which estimate QoE from MAC-level QoS. Furthermore, we evaluate the effect of content types on the QoE by using estimated values from MAC-level QoS.

This paper deals with audio-video transmission over IP networks and proposes a method of estimating perceptual QoS (i.e., user-level QoS) in real time in terms of the interval scale, which is referred to as the psychological scale. The proposed scheme utilizes both temporal and no reference spatial (picture's) quality, which are application-level QoS, and estimates the user-level QoS from the application-level QoS. To evaluate the effectiveness of the proposed scheme, we compare the proposed scheme with a scheme which uses only temporal quality for the estimation and a scheme with both temporal and Full Reference spatial quality. We made a simple experiment of audio-video IP transmission; we measured the application-level QoS directly and assessed the user-level QoS by the method of successive categories. Applying the principal component analysis and multiple regression analysis to the experimental results, we obtained multiple regression lines (i.e., equations for the estimation). As a result, we see that the user-level QoS can be estimated with high accuracy by using both temporal quality and spatial quality.

This paper studies packet scheduling schemes for audio-video transmission in a wireless LAN with HCF controlled channel access (HCCA) of the IEEE 802.11e MAC protocol. We first propose two kinds of scheduling schemes which are referred to as the static scheduling (SS) and multimedia priority dynamic scheduling (MPDS). In the SS scheme, the hybrid coordinator (HC) allocates the duration of transmission opportunity (TXOP) for each station statically. The TXOP duration in the SS scheme can be shorter or longer than that in the reference scheduler of the IEEE 802.11e standard. On the other hand, in the MPDS scheme, the HC first allocates the duration of TXOP to each station according to the SS scheme; it then gives additional TXOP duration to stations which had audio or video packets in their source buffers at the end of the previous TXOP. We then compare the SS scheme, the MPDS scheme, and the reference scheduler in terms of application-level QoS and user- level QoS (perceptual QoS). Numerical results show that the SS and MPDS schemes can accommodate a larger number of multimedia stations in a cell than that of the reference scheduler under the condition that user-level QoS is kept high.

This paper studies a multipoint-to-multipoint TV conferencing application over IP networks and assesses its user-level QoS with two types of QoS mapping. In utilizing the application, the user perceives quality of the communication with every other conferee; we refer to the quality as individual user-level QoS to a conferee. According to the individual user-level QoS, he/she totally judges the quality of the application, which is refer to as overall user-level QoS for the user. The overall user-level QoS of the application can be affected by the individual one to each conferee; therefore, it is difficult to clarify QoS parameters which affect the overall user-level QoS. This paper tackles the problem by utilizing two types of QoS mapping: mapping between the two kinds of user-level QoS and that between user-level QoS and application-level QoS. In this paper, an experiment with a simple task by three conferees was carried out. The user-level QoS is assessed by one of the psychometric methods. As a result of the two types of QoS mapping, we find two interesting results. First, when a user communicates with the other two conferees, the lower individual user-level QoS has more effect on the overall user-level QoS than the higher one. Second, the individual user-level QoS can depend on not only its application-level QoS but also that of the other conferees.

This paper proposes a session control method which provides a mechanism to achieve desirable user-level QoS (i.e., perceptual QoS) of audio-video transmission over IP networks. The proposed method, which is referred to as GPSQ (Guarantee of Psychologically Scaled Quality), utilizes a user-level QoS parameter of the interval scale in the psychometric analysis. GPSQ involves four kinds of components: media terminals, a SIP server, a QoS manager and bandwidth-controllable routers. From the media terminals, the SIP server obtains the information for this control, which is further delivered to the QoS manager. The QoS manager keeps a database of representative regression lines which express user-level QoS as a function of guaranteed bandwidth of audio and that of video. Using the regression line, the QoS manager calculates necessary guaranteed bandwidth according to user-level QoS specified by the user. Then, the QoS manager sets up routers to guarantee the calculated bandwidth. We implemented GPSQ in a simple experimental network and confirmed its effectiveness.

This paper studies a method of utilizing the cross- modal influences of audio and video, which is referred to as the mutually compensatory property here, in order to enhance user- level QoS (i.e., perceptual QoS) in IP networks. For that purpose, we clarify how the mutually compensatory property depends on the content type. We conducted an experiment on audio-video transmission over a simple IP network with ten contents which are classified into five types. In the experiment, we transmitted CBR audio-video streams with various values of guaranteed bandwidth for audio and that for video under the condition that the total guaranteed bandwidth of the two streams is constant. Then, we assess user-level QoS and perform QoS mapping between user-level and application-level by multiple regression analysis. Using the obtained regression lines, we examine the dependence of the property on the content type. As a result, we confirm that the mutual compensation relationship between audio and video can be classified into three kinds. Furthermore, we investigate how we should allocate the bandwidth between audio and video according to the content type in order to maximize the user-level QoS.

This paper proposes a method for testing feasibility of QoS control based on QoS mapping. We focus on the issue in wireless IP networks with IEEE 802.11 MAC. In wireless multimedia applications, it is important to investigate how we enhance user-level QoS of audio-video transmission. As one of the methods, we can control QoS at lower levels so that user-level QoS keeps high by means of QoS mapping. However, there is no guarantee that a given scheme of QoS control at lower-levels can give desirable user-level QoS. Therefore, we must examine the feasibility of the QoS control. In order to test it, we introduce concepts of the feasibility space and the decision boundary, which are obtained by the QoS mapping with multiple regression analysis. To show the effectiveness of our method, we utilize the simulation results in a previous work by the authors' research group. The simulation treats application-level QoS control and MAC-level one; we adopt EDCA of IEEE 802.11e as MAC-level control.

This paper proposes a real-time estimation method of perceptual QoS (i.e., user-level QoS) of audio-video transmission over IP networks. The estimated quality reflects cross-modal influences of audio and video in the framework of ITU-T Recommendation J. 148. As objective measures of audio and video for the use of the perceptual QoS estimation, we select application-level QoS parameters for media synchronization quality. The perceptual quality is expressed in terms of the user-level QoS parameter of the interval scale, which is referred to as the psychological scale. As the multimedia quality integration function, we propose QoS parameter mapping between the application-level and the user-level. We made subjective quality assessment of simulated audio-video transmission over an IP network by the rating-scale method. We then calculate the psychological scale, utilizing the law of categorical judgment. Furthermore, we obtain the QoS mapping relations by the principal component analysis and multiple regression analysis. The regression formula thus derived can estimate the psychological scale with high accuracy unless the video frame rate is low. Finally we present an extension of the proposed method to continuous time estimation of user-level QoS.

This paper proposes a link quality-based hybrid routing (LQHR) protocol for wireless ad hoc networks. LQHR takes account of link quality such as SNR between adjacent nodes and the link utilization level of each node. In LQHR, each node maintains routing information produced by OLSR, which is a proactive routing protocol for ad hoc networks. When a source makes a communication request, it finds a route to the destination node on the basis of the link quality. We implemented LQHR on an experimental network, where laptop PCs were arranged on the same floor of a building, and audio-video streams were transmitted along a route selected by LQHR. To evaluate the effectiveness of LQHR in audio-video transmission, we compared application-level QoS of LQHR with that of the standard OLSR in the presence of another traffic flow in the network. As a result, we found that the routes selected by LQHR can provide shorter transmission delays and lower loss ratios in the transmission of the audio-video streams than those selected by the standard OLSR

This paper assesses application-level QoS and user-level QoS (i.e., perceptual QoS) in the case where audio and video streams are transferred with the EDCA (enhanced distributed channel access) of the IEEE 802.11e MAC. In the EDCA, a station can transmit multiple MAC frames during a TXOP (transmission opportunity); this is referred to as TXOP-bursting or CFB (contention free bursting). By simulation, we first examine the effect of the TXOP-bursting on application-level QoS in the presence of transmission errors and associated retransmission of MAC frames. Simulation results show that the TXOP-bursting can improve average video MU (media unit) delay, MU loss ratio, and media synchronization quality, particularly when the AP (access point) sends audio and video streams to stations in the downlink direction. Furthermore, we assess user-level QoS by the method of successive categories, which is a psychometric method. Numerical results show that the TXOP-bursting can also improve the user-level QoS

This paper investigates user-level (or perceptual) QoS of audio-video transmission over IP networks with terminals of different size monitors. As the first step of our research, we treat large-sized monitors. By experiment, we assess the user-level QoS for three monitor sizes (80, 60 and 40 inches) and perform QoS mapping. In our experiment, we changed the monitor size by using a projector. The user-level QoS is assessed by the method of successive categories, which is one of the psychometric methods. Moreover, we performed QoS mapping from application-level to user-level with multiple regression analysis. The experimental result indicates that when the degradation of the quality of the audio-video transmission is not noticeable, the user-level QoS of the large monitor size (i.e., 80 inch) becomes the highest. We also observe that when we can notice a low level of degradation of the transmission, the user-level QoS parameter of the large monitor size becomes the smallest. When the degradation is too much, we hardly notice any difference in the user-level QoS between the monitor sizes. We thus confirmed that the monitor size is one of important application-level QoS parameters we must consider

This paper assesses application-level QoS of live audio and video multicasting in a wireless ad hoc network, focusing on the quality of media synchronization. In particular, we investigate the inter-destination synchronization quality of three schemes: the master-slave destination scheme, the synchronization maestro scheme, and the distributed control scheme. The inter-destination synchronization adjusts the output timing among destinations in a multicast group. We perform computer simulation with a grid topology network of IEEE 802.11b. In the simulation, a source terminal multicasts a pair of audio and video streams to six destinations by using ODMRP (on-demand multicast routing protocol), which is a multicast routing protocol for ad hoc networks. We assess the application-level QoS when the source terminal moves. From the simulation results, we investigate the achievable application-level QoS for the three inter-destination synchronization schemes

This paper studies audio-video transmission in a wireless LAN with the EDCA of the IEEE 802.11e MAC protocol in terms of application-level QoS and user-level QoS (i.e., perceptual QoS). By simulation, we first compare the performance of the EDCA and the legacy IEEE 802.11 DCF in the case where audio, video, and data are transmitted in a common wireless channel. The EDCA supports priority control for multimedia transmission. Numerical results show that the EDCA can improve the audio and video quality of the DCF when data traffic load is heavy. We then study the effects of dynamic video-resolution and media synchronization control on the application-level QoS. Simulation results show that the two kinds of control at the application layer in addition to priority control with the EDCA at the MAC layer is effective in achieving excellent application-level QoS of audio-video transmission. Furthermore, we assess user-level QoS by the method of successive categories, which is one of the psychometric methods.

This paper proposes a scheme for quantitative assessment of user-level (or perceptual) quality of service (QoS) for audio-video transmission by means of two psychometric methods: the method of paired comparisons and the law of comparative judgment. Moreover, we discuss QoS mapping from application-level QoS to user-level QoS by principal component analysis and multiple regression analysis. In the assessment, we simulate the transmission of an audio-video stream over a loaded network. In order to investigate the effect of the contents on QoS mapping, we treat two types of audio-video streams. By experiment, we demonstrate that our scheme can construct an interval scale as the user-level QoS parameter for each stream and represent it as a function of two application-level QoS parameters with high accuracy. We notice that the multiple regression line depends on the contents. We also propose the concept of control gain by media synchronization, which indicates how much media synchronization control subjectively lightens the average network load.

This paper investigates the effect of the mean and jitter of end-to-end delay on use-level QoS of live audio-video transmission. The temporal structure of continuous media can be disturbed by the delay jitter of packets. The temporal disturbance causes a subjective degradation of the media. On the other hand, long end-to-end delay also degrades the quality of live media subjectively. Therefore, it is necessary to study the effect of the mean delay and the delay jitter on user-level QoS degradation. In this paper, we assess the effect by experiment and a psychometric method. In our experiment, we actually delay the packets of live audio and video according to the Pareto-normal distribution with a network emulator and assess user-level QoS of the live transmission with the method of successive categories, which is a psychometric method. As a result, we show that the standard deviation of' delay is more dominant than the mean delay in the effect on the user-level QoS. We also perform QoS mapping from application-level to user-level with multiple regression analysis. From the QoS mapping, we clarify the effect of degradation of the application-level QoS caused by the mean and standard deviation of delay on the user-level QoS.

This paper assesses the application-level QoS of live audio and video multicasting in a wireless ad hoc network. We regard the quality of media synchronization as the major part of the application-level QoS. We also investigate the inter-destination synchronization quality of three schemes: the master-slave destination scheme, the synchronization maestro scheme, and the distributed control scheme. The inter-destination synchronization adjusts the output timing among destinations in a multicast group. We perform computer simulation with a string topology network. In the simulation, a source terminal multicasts the media streams to six destinations by using ODMRP (on-demand multicast routing protocol), which is a multicast routing protocol for ad hoc networks. The simulation result shows that the master-slave destination scheme, which sends smaller numbers of control packets for inter-destination synchronization control than the other two schemes, provides higher quality of application-level QoS; however, the QoS of this scheme is sensitive to the location of the master destination. When it is difficult to find an appropriate destination for the master, the synchronization maestro scheme can be the second best choice.

This work compares the application-level QoS of four schemes for live audio and video streaming in a wireless ad hoc network by simulation. There are two approaches to audio-video transmission: single-stream and multi-stream. The former transmits a single transport stream of interleaved audio and video, while the latter treats the two media as separate transport streams. Each approach has an alternative of whether media synchronization control at the destination is carried out or not. Thus, we have the four basic schemes. In the simulation, we also assess the network-level QoS. We then show that the multi-stream approach with media synchronization control is the best scheme among the four schemes.

This paper addresses an inter-stream synchronization issue between haptic media and voice in networked virtual environments. The paper proposes a media synchronization algorithm for the two types of media stream by enhancing the virtual-time rendering (VTR) algorithm, which the authors previously proposed. The new algorithm employs two types of error range in order to keep high quality of intra-stream synchronization at the expense of slight deterioration in the inter-stream synchronization quality. One is the imperceptible range of inter-stream synchronization error, and the other is the allowable range of the error. If the error is beyond the allowable range, the algorithm tries to reduce the error gradually until the error becomes within the imperceptible range. Otherwise, it accepts the error. By carrying out an experiment in which two users do collaborative work with haptic media and voice in a virtual space, we demonstrate the effectiveness of the algorithm quantitatively.

The paper presents an application-level QoS comparison of three inter-destination synchronization schemes: the master-slave destination scheme; the synchronization maestro scheme; the distributed control scheme. The inter-destination synchronization adjusts the output timing among destinations in a multicast group for live audio and video streaming over the Internet/intranets. We compare the application-level QoS of these schemes by simulation, in which we assume an intranet. From the comparison, we clarify their features and find the best scheme in the environment. The simulation result shows that the best scheme depends on the network configurations and the acceptable inter-destination synchronization quality of the applications.

This paper assesses application-level QoS of both web access and voice-video streaming services on the Internet. We investigate how AFCP values in the two services affects their application-level QoS for various network scales, which are represented by the end-to-end delay between the Web server and the client. As a result of the experiment, we see that we can keep higher application-level QoS of the two services if the Web access service is offered as a different class from that of the streaming service or in the same class as that of the video flows with the video-flows' higher drop precedence. Furthermore, as the end-to-end delay increases, the effect of the choice of the AFCP values decreases. This implies that in such a network environment we do not need to give much consideration to AFCP marking.

This paper proposes two group synchronization schemes, which adjust the output timing among destinations, in networked virtual environments where voice and video streams are output in a 3-D virtual space. We introduce the concept of the global importance of each media object, which includes a voice stream and a video one. The global importance is determined by the positional relations among media objects, and it represents how strongly each object attracts observer's attention in the virtual space. We enhance a centralized group synchronization control scheme called the synchronization maestro scheme and a distributed control scheme, which were previously proposed by the authors, in order to adjust the output timing of media streams among destinations according to the global importance. In addition, we illustrate the effectiveness of the proposed schemes by experiment.

This paper quantitatively evaluates the mutually compensatory property in audio-video transmission over the Internet by the psychometric method. In our previous study, we used MOS (mean opinion score) as the user-level QoS parameter; however, MOS is an ordinal scale, and therefore, its value does not necessarily represent exact difference between the observers sensory magnitude. In order to solve this problem, we construct an interval scale from the MOS values, utilizing the law of categorical judgment. We then examine how closely MOS approximates an interval scale. Furthermore, we obtain QoS mapping relations between application-level and user-level by the principal component analysis and multiple regression analysis. The regression formula thus derived can demonstrate how much one medium compensates for the other in a quantitative way with the interval scale.

This paper assesses the media synchronization quality of audio-video transmission over the IEEE 802.11b wireless LAN under Bluetooth interference by experiment. In this situation, the temporal constraints of audio and video may be disturbed, since the Bluetooth interference cause delay jitter due to retransmission and carrier sensing in the MAC layer of IEEE 802.11b. In the experiment, a media server transfers stored video and audio streams to a mobile terminal over an IEEE 802.11b wireless LAN; as radio interference from Bluetooth, two other terminals transmit data over Bluetooth at the same time. We examine the influence of Bluetooth on the audio-video transmission over IEEE 802.11b in terms of the Bluetooth signal level and its data load. We also apply a media synchronization control scheme and confirm its effectiveness.

This paper presents experimental results of inter-destination synchronization control in an integrated wired and wireless network. The experimental system consists of interconnected wired and wireless networks; the former is a router-based IP network, and the latter is a wireless LAN with two base stations. A server in the wired network multicasts a pair of audio and video streams to a terminal in the same wired network and a wireless terminal, which performs handover from one base station to the other. We exert inter-destination synchronization control between the two terminals using the synchronization maestro scheme, which the authors previously proposed. In the experiment, we measured the synchronization quality of inter-destination as well as intra-stream and inter-stream when handover occurred, and we confirmed the effectiveness of the control.

This paper studies user-level (or perceptual) QoS for multimedia applications over the Internet and demonstrates that media can compensate for each other from a perceptual point of view. For this study, we formulate the problem of the mutually compensatory property in terms of QoS mapping relation between application-level and user-level; we then conduct a simple experiment on lip synchronization to investigate the property. As a result, we observe the existence of the mutually compensatory property between voice and video. It is also shown that the property exists within a limited range of combinations of application-level QoS parameters. Furthermore, we point out the advantage of QoS control utilizing the property.

This paper proposes a distributed control scheme for causality and media synchronization in networked games where plural media streams such as computer data, voice, and video are dealt with together. According to the network load, the scheme adaptively preserves the temporal and causal relationships which exist among the media streams. It also performs group (or inter-destination) synchronization control, which adjusts the output timing among terminals for keeping the fairness among them. Furthermore, it exerts error control over computer data and video traffic control. We demonstrate the effectiveness of the scheme by carrying out an experiment of a networked shooting game.

This paper examines the effect of segmentation mismatch on audio-video transmission by Bluetooth. We focus on the segmentation mismatch caused by the difference between the RFCOMM maximum frame size and the baseband packet payload size. By experiment, we assessed the maximum throughput and media synchronization quality for various types of ACL packets. In the experiment, a media server transferred stored video and audio streams to a single terminal with point-to-point communication; we supposed no fading environment and added white noise by which interference from DSSS systems is modeled. The experiment showed that the effect of segmentation mismatch is large especially when the total bit rate of the two streams is near the channel transmission rate. We also observed that the media synchronization control is effective in compensating for the disturbance by the segmentation mismatch in noisy environments.

This paper assesses the media synchronization quality in a Bluetooth LAN access system by experiment. We consider a situation where a server transfers stored video and audio streams to a terminal with point-to-point communication. Both FEC and ARQ are adopted as error control schemes. The slide control scheme is used for media synchronization control. We examine how the interference from DSSS systems affects the quality. For simplicity of experiment, we suppose that the Bluetooth terminal does not move while the media streams are transferred, i.e., fading does not occur. In the experiment, we evaluated several performance measures such as mean square error of intra-stream synchronization. We then found that delay jitters caused by retransmission of DM1 packets disturbed temporal relations of the streams. We also observed that the media synchronization control is effective in compensating for the disturbance.

This paper proposes two kinds of priority scheduling schemes, which are referred to as priority scheme I and priority scheme II, for multimedia transmission with the PCF (point coordination function) of the IEEE 802.11 standard MAC protocol. This paper focuses on a single BSA (basic service area) in an infrastructure network and adopts a hierarchical scheduling architecture. In the architecture, the AP (access point) schedules wireless terminals, and each wireless terminal schedules voice, video and data sources. In priority scheme I, wireless terminals exert priority control where voice and video transmission is given higher priority over data transmission. On the other hand, priority scheme II carries out priority control at both AP and wireless terminals. By simulation, we compare the performance of priority scheme I, priority scheme II and a non-priority scheme. Numerical results show that these priority schemes can support a larger number of multimedia wireless terminals than the non-priority scheme under the same condition.

This paper proposes a media synchronization scheme with adaptive /spl Delta/-causality control for networked multimedia applications in which the temporal and causal relationships exist among media streams such as computer data, voice, and video. In the scheme, the adaptive /spl Delta/-causality control, which varies the value of /spl Delta/ of the /spl Delta/-causality control according to the network load, is performed for causality, and the virtual-time rendering (VTR) algorithm, which the authors previously proposed, is used for media synchronization. The scheme also performs group (or inter-destination) synchronization control, which adjusts the output timing among terminals for preservation of the fairness among them, as well as error control and traffic control. We demonstrate the effectiveness of the scheme by carrying out an experiment of a networked shooting game.

This paper studies effect of packet scheduling algorithms at routers on media synchronization quality in live audio and video transmission by experiment. In the experiment, we deal with four packet scheduling algorithms: first-in first-out, priority queueing, class-based queueing and weighted fair queueing. We assess the synchronization quality of both intra-stream and inter-stream with and without media synchronization control. The paper clarifies the features of each algorithm from a media synchronization point of view. A comparison of the experimental results shows that weighted fair queueing is the most efficient packet scheduling algorithm for continuous media among the four.

This paper assesses the quality of media synchronization of recovery control schemes from asynchrony, which are referred to as reactive control schemes here, in terms of objective and subjective measures. We deal with four reactive control techniques: skipping, discarding, shortening and extension of output duration, and virtual time-contraction and time-expansion. We have carried out subjective and objective assessment of the media synchronization quality of nine schemes, which consist of combinations of the four techniques. The paper makes a comparison of media synchronization quality among the schemes. It also clarifies the relations between the two kinds of quality measures.

By experiment, this paper studies the responsiveness of layered multicast and feedback control for video traffic in reacting to network congestion over the Internet. We examine the influence of the responsiveness on the performance in our lip synchronization experimental system, where video and voice streams are multicast under controllable traffic and delay conditions. The paper also demonstrates the possibility of performance improvement by using the two techniques together. This scheme tries to solve the problem of large leave latency in layered multicast by taking advantage of the quickness in feedback control.

This paper proposes a transport-layer error recovery scheme using retransmission for live audio-video streams transferred over QoS non-guaranteed networks. The scheme employs an enhanced version of the virtual-time rendering (VTR) media synchronization algorithm, which adjusts media rendering-time according to the network condition and is referred to as RVTR (retransmission with VTR). By an experiment, we assess the synchronization quality of both intra-stream and inter-stream in RVTR and four other schemes for audio-video transmission. A comparison of the experimental results indicates that joint application of retransmission control and the VTR synchronization control as in RVTR is effective in the improvement of media synchronization quality and that either control alone does not produce much improvement.

This paper considers a networked multimedia application like distance learning in which a tutor distributes a stored video stream to students, and he/she sometimes explains the contents of the stored video with live voice and video streams. We propose a joint synchronization mechanism among the media streams. The mechanism takes advantage of the difference in the features of live and stored media. It makes the buffering time of the stored video stream at the destination longer than that of the live media streams. We also demonstrate the effectiveness of the mechanism by experiment. Furthermore, we examine the influence of the buffering time of the stored video stream on performance.

This paper presents results of an experiment in which plural video and voice streams transmitted over a network are output synchronously in networked virtual environments. We propose an adaptive QoS (quality of service) control scheme for video and voice traffic. According to the importance of each media object, CPU load and network load, the scheme adjusts the rendition rate of a 3D virtual space and the video resolution as well as the accuracy of media synchronization; also it can display a CG object instead of a less-important video object. In addition, the paper quantitatively examines the effect of the adaptive QoS control on the performance.

This paper makes a survey of algorithms proposed for continuous media synchronization in network environments. We classify media synchronization control techniques used in the algorithms into four categories: basic control, preventive control, reactive control and common control. We also pick four representatives from among the media synchronization algorithms and give outlines of them in order to explain how the techniques are employed in each algorithm. Furthermore, we make a comparison among 38 algorithms in terms of factors such as clocks, advance information on network delay bounds and synchronization control techniques, which determine the design of each algorithm.

For networked multimedia applications like distance learning in which a tutor explains the contents of a stored video stream to students with live voice and video streams, this paper proposes a joint synchronization mechanism between live and stored media. The mechanism makes the buffering time of stored media at each destination longer than that of live media. It also carries out retransmission control over stored media and group synchronization control, which adjusts the output timing among destinations in a multicast group. By experiment, we demonstrate the effectiveness of the mechanism.

This paper proposes a bandwidth allocation scheme where priority-based control is enhanced for call acceptance and bandwidth allocation among multimedia component calls. The bandwidth allocation schemes previously proposed by the authors perform the elimination of component calls and the deferment of bandwidth allocation to them according to their quality-of-services (QoSs) and importance for communications. The proposed scheme in this paper further keeps a certain amount of the bandwidth reserved for indispensable component calls for communications. By computer simulation, we evaluate the performance such as call dropping probability when a handoff occurs and show the effectiveness of the proposed scheme.

This paper proposes a media synchronization scheme with causality control for distributed multimedia applications in which the temporal and causal relationships exist among media streams such as computer data, voice and video. In the scheme, the /spl Delta/-causality control is performed for causality, and the virtual-time rendering (VTR) algorithm, which the authors previously proposed, is used for media synchronization. The paper deals with a networked shooting game as an example of such applications and demonstrates the effectiveness of the scheme by experiment.

This paper presents a performance comparison between the single-stream and the multi-stream approaches for lip synchronization of live media (voice and video). The former transmits a single transport stream of interleaved voice and video, while the latter treats the two media as separate transport streams. Each approach has an option not to exert the synchronization control at the destination, which leads to four basic schemes. On an interconnected ATM-wireless LAN, we implemented the four basic schemes with RTP/RTCP on top of UDP and two variants which exercise dynamic resolution control of JPEG video. By the performance measurement of the six schemes, we compare them to identify and evaluate the advantages and disadvantages of each approach.

This paper makes an experimental comparison of performance among six media synchronization schemes in collaborative systems, where drawing on a shared window, voice and video streams need to be synchronized with each other. Five of the six schemes are based on the virtual-time rendering (VTR) algorithm proposed by the authors, and they differ from each other in the relation between master and slave streams. The remaining one employs another algorithm. In our experimental system using an interconnected ATM-wireless LAN, live drawing voice and video input at a source workstation are transferred separately and then they are synchronized and output at a destination workstation. This paper also presents the system performance such as mean square error of inter-stream synchronization and average delay time. In addition, we examine the effect of intrastream and inter-stream synchronisation control on the performance.

This paper proposes a group synchronization mechanism, which synchronizes slave destinations with the master destination, for stored media in multicast communications. At the master and slave destinations, intra-stream and inter-stream synchronization mechanisms which were proposed by the authors are employed to output the master media stream and slave media streams synchronously. We achieve group synchronization by adjusting the output timing of the master media stream at each slave destination to that at the master destination. We also deal with traffic control by media scaling and control of joining an in-progress multicast group. Furthermore, the paper presents experimental results using an ATM network. It shows the validity of the mechanism and illustrates the influence of parameters on the system performance.

This paper proposes a scheme for synchronization of stored video and audio streams in the Personal Handy phone System (PHS). A video stream of H.263 is transmitted over a PHS channel with ARQ control, while an audio stream of 32 kbit/s ADPCM is sent on another channel without any control. In order to preserve the temporal constraints within the video stream as well as the relationship between the video and audio streams, we adopt a new control scheme which modifies the target output time according to the amount of video data in the receive buffer. Through simulation we assess the characteristics of this scheme in both random and burst error environments and confirm the effectiveness of the scheme.

This paper describes an experiment of a temporal resolution control scheme of stored video traffic in a wireless LAN. The scheme switches the resolution between two levels according to the network load. We propose a simple method of detecting network congestion which uses only local information. We have implemented the scheme on a wireless LAN based on ALTAIR and have made performance measurement in terms of throughput and total output time. As a result, we observe that although the scheme does not improve the throughput performance, it is effective in suppressing an increase of the total output time due to the network delays.

The author makes a performance comparison between a connection-oriented (CO) logical link control (LLC) protocol and a connectionless (CL) LLC protocol in a high-speed satellite packet network with the time-division-multiple-access (TDMA)-reservation multiple-access protocol. In particular, the way that the channel transmission rate affects the performance of the two LLC protocols is examined. The most suitable LLC protocol for any given system environment is determined. The comparison indicates that the CL-LLC protocol has more advantages over the CO-LLC protocol as the channel transmission rate increases and that the suitable region for the CL-LLC protocol becomes wider as the receiver-buffer capacity increases and as the network load decreases.>

A performance analysis of slotted rings is presented based on a register-insertion method which is proposed for broadband ring topology networks. The analysis is carried out under the assumption that each node contains a finite buffer space. A Markovian model is developed for the ring. The arrival process of packets from upstream nodes is determined from the model. The probability of buffer overflow and the probability distribution of the number of packets in a buffer are calculated using the model. The analytical results are shown to be accurate by comparing them with simulation results.>

A performance comparison is made between a connection-oriented (CO) logical link control (LLC) protocol and a connectionless (CL) LLC protocol in a high-speed satellite packet network with the TDMA-reservation multiple access protocol. In particular, the authors examine how the channel transmission rate affects the performance of the two LLC protocols and then determine which LLC protocol is more suitable for any given system environment. The comparison indicates that the CL-LLC protocol has more advantages over the CO-LLC protocol as the channel transmission rate increases and that the suitable region for the CL-LLC protocol becomes wider as the receive-buffer capacity increases and as the network load decreases.>

In packet broadcast networks with buffered users, queues of packets at the users interfere with each other through a shared broadcast channel. Therefore, it is difficult to analyze the performance of such Systems by classical queueing theory. This paper provides an approximate performance analysis of a slotted nonpersistent CSMA-CD system with a finite number of users, each having a buffer of finite capacity. We develop an approximate Markovian model of the system with a multidimensional state vector. To analyze the model, We utilize an approximate analytical technique called equilibrium point analysis (EPA). Then, the throughput and average packet delay characteristics are obtained and the system stability behavior is demonstrated. An approximate expression is also derived for the probability distribution of the number of packets in a buffer. Numerical results from both analysis and simulation are given to illustrate the accuracy of those analytic results. Using the analytic results, we examine the effect of the collision detection capability on the system performance. Furthermore, we consider how to select both the rescheduling probability and the buffer size that guarantee the system stability, keep the probability of buffer overflow sufficiently small, and at the same time minimize the average packet delay.

SRUC is an adaptive multiple access protocol which combines the slotted ALOHA protocol and a reservation protocol; it switches from one to the other according to the state of the channel. This paper studies SRUC with a go-back-NARQ scheme (SRUC/GBN) in a satellite communication environment to improve the performance over SRUC with a stop-and-wait ARQ scheme (SRUC/SAW) previously studied. The performance of SRUC/GBN is analyzed in this paper, it is assumed that each user handles one message at a time and that the number of packets in a message is geometrically distributed. An approximate Markovian model for SRUC/GBN is developed which has a multidimensional state vector. The model is analyzed by the technique of equilibrium point analysis. The throughput and average message delay performance is derived. Numerical results from both analysis and simulation are given. The performance of SRUC/GBN is also compared with that of SRUC/SAW and other multiple access protocols.

The dynamic behavior of a reservation system with a slotted ALOHA reservation channel is analyzed in this paper. It is assumed that each user handles one message at a time and that each message consists of a group of packets with a general probability distribution for group size. An approximate Markovian model of the system is developed on the assumption that the state transition can occur only at the beginning of each frame. The model is analyzed by an approximate analytical technique called equilibrium point analysis. The throughput and average message delay characteristics are obtained, and the system stability behavior is demonstrated. A procedure is also given to get the optimum set of system parameters under constraint that the system is stable.

The dynamic behavior of the R-ALOHA packet broadcast system with multipacket messages is analyzed in this paper. It is assumed that each user handles one message at a time and the number of packets in a message is geometrically distributed. A Markovian model of the system is first formulated which explicitly contains the influence of the propagation delay of the broadcast channel. An approximate technique called equilibrium point analysis (EPA) is utilized to analyze the multidimensional Markov chain. The system stability behavior and the throughput-average message delay performance are demonstrated by the EPA. Numerical results from both analysis and simulation are given to assess the accuracy of the analytic results. Applying the analytic results to the slotted ALOHA with single packet messages, we prove mathematically that a method by Kleinrock and Lam for taking into account the influence of the propagation delay is an excellent approximation.