Interactive audio-video applications over IP networks have subjective tradeoffs between fidelity and latency owing to packet buffering at the receiver. Increasing the buffering time improves the fidelity, whereas it degrades the latency. This paper makes the subjective tradeoff between fidelity and latency clear in a quantitative way. In addition, we examine the effect of tasks on the subjective tradeoff. In evaluating the effect of tasks, we use two tasks according to ITU-T Recommendation P.920. An experiment was conducted to measure user-level QoS of an interactive application with the psychometric methods. We then investigate the subjective tradeoff quantitatively by QoS mapping. The experimental results confirm that there exists the buffering time which makes user-level QoS the highest. The results also show that the optimum buffering time depends on the kind of task.

This paper studies packet scheduling schemes with QoE (user-level QoS) guarantee for audio-video transmission in a wireless LAN with HCF controlled channel access (HCCA) of the IEEE 802.11e MAC protocol. We first propose the static scheduling (SS) scheme, which grants adjustable transmission opportunity (TXOP) duration for constant bit rate (CBR) traffic. The SS scheme can determine the minimum TXOP duration capable of guaranteeing high QoE; it can maximize the number of admitted flows. As the burstiness of variable bit rate (VBR) traffic cannot be absorbed by the SS scheme, we also propose the multimedia priority dynamic scheduling (MPDS) scheme, which can absorb the burstiness through allocating additional TXOP duration. We then compare the SS scheme, the MPDS scheme, and the reference scheduler (TGe scheme) in terms of application-level QoS and user-level QoS (QoE). Numerical results show that in the SS scheme, the QoE can be kept relatively higher even when the TXOP duration is reduced in the case of video with the I picture pattern; this implies that more flows can be admitted. In the case of video with the IPPPPP picture pattern, which has the VBR characteristic more remarkably, reducing the TXOP duration according to the SS scheme will deteriorate the QoS level. In this case, the MPDS scheme performs better when the number of multimedia stations is small. However, the performance of the MPDS scheme deteriorates with the increase of the number of multimedia stations, though the results are comparable to or even better than those of the SS and TGe schemes.

This paper studies congestion control schemes for integrated variable bit-rate (VBR) video and data communications, where the quality of service (QOS) of each medium needs to be satisfied. In order to control congestion, we exert here either dynamic resolution control or QOS control. The dynamic resolution control scheme in this paper dynamically changes the temporal or spatial resolution of video according to the network loads. The QOS control scheme here assigns a constant capacity of buffer to each connection and determines the video resolution in order to guarantee the QOS of each medium at the connection establishment. The performance of these schemes is evaluated through simulation in terms of throughput, video frame delay probability distribution, and video frame loss rate. We also examine the effects of priority scheduling and packet discarding on the performance. Numerical results indicate that both dynamic resolution and QOS control attain low delay jitters as well as large video and data throughput. In particular, the QOS control is shown to be more suitable for integrated VBR video and data communications.

This paper studies a set of lip synchronization mechanisms for heterogeneous network environments. The set consists of four schemes, types 0 through 3, which are classified into the single-stream approach and the multi-stream approach. Types 0 and 1 belong to the single-stream approach, which interleaves voice and video to form a single transport stream for transmission. On the other hand, types 2 and 3, both of which are the multi-stream approach, set up separate transport streams for the individual media streams. Types 0 and 2 do not exert synchronization control at the destination, while types 1 and 3 do. We first discuss the features of each type in terms of networks intended for use, required synchronization quality of each medium, physical locations of media sources and implementation complexity. Then, a synchronization algorithm, which is referred to as the virtual-time rendering (VTR) algorithm, is specified for stored media; MPEG video and voice are considered in this paper. We implemented the four types on an ATM LAN and on an interconnected ATM-wireless LAN under the TCP protocol. The mean square error of synchronization, total pause time, throughput and total output time were measured in each of the two networks. We compare the measured performance among the four types to find out which one is the most suitable for a given condition of the underlying communication network and traffic.

This paper proposes a media synchronization mechanism for live media streams. The mechanism can also handle stored media streams by changing parameter values. The authors have implemented the mechanism on a lip-synch experimental system. Live video and voice streams input at a source workstation are transferred, and then they are synchronized and output at a destination workstation. This paper also evaluates the system performance such as mean square error of synchronization, average output rate, and average delay.

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

This letter analyzes the performance of a virtual-token passing scheme, fair BRAM, for local area networks. A Markovian model of the system is developed and analyzed by the equilibrium point analysis. The throughput and average message delay characteristics are shown.

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 studies a congestion control scheme in integrated variable bit-rate video, audio and data (e. g. , image or text) communications, where each video stream is synchronized with the corresponding audio stream. When the audio and video streams are output, media synchronization control is performed. To avoid congestion, we employ a dynamic video resolution control scheme which dynamically changes the video encoding rate according to the network loads. By simulation, the paper evaluates the performance of the scheme in terms of throughput, loss rate, average delay, and mean square error of synchronization. Numerical results show the effectiveness of the scheme.

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 proposes a bandwidth allocation scheme which improves degradation of communication quality due to handoffs in mobile multimedia networks. In general, a multimedia call consists of several component calls. For example, a video phone call consists of a voice call and a video call. In realistic environments, each component call included in one multimedia call may have different requirements for quality-of-service (QoS) from each other, and priorities among these component calls often exist with respect to importance for communications. When the available bandwidth is not enough for a handoff call, the proposed scheme eliminates a low priority component call and defers bandwidth allocation for a component call whose delay related QoS is not strict. Moreover, in the allocation, the scheme gives priority to new calls and handoff calls over a deferred call and also performs bandwidth reallocation to eliminated component calls. By computer simulation, we evaluate the performance such as call dropping probability and show effectiveness of the proposed scheme.

In this paper, we propose a framework for the real-time estimation of a multidimensional QoE of Multi-View Video and Audio (MVV-A) IP transmission. The framework utilizes linear multiple regression analysis with application-level and transport-level QoS parameters which can be measured in real time. In order to cope with a variety of MVV-A usage-situations, we introduce the concept of usage-situation type for grouping usage-situations with similar features to apply a representative regression line. We deal with two contents, two camera arrangements, and two user interfaces for viewpoint change as representative examples of the usage-situations. We assess multidimensional QoE of MVV-A with various types of average load, playout buffering time, and delay in the network. We then conduct the multiple regression analysis for the multidimensional QoE values represented by a psychological scale. 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 assesses the media synchronization quality 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 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.

A performance comparison between TCP and UDP in PHS Internet access is made by experiment from a media synchronization point of view. We consider a situation where PHS mobile terminals access H. 263 video and G. 726 audio stored at a media server by a streaming method. PIAFS is adopted as the data link protocol for the PHS wireless channels. We examined how white noise and Rayleigh fading on the PHS channel as well as the Internet traffic affect the performance. For the comparison, we evaluated several performance measures such as the coefficient of variation of output interval, and found that UDP outperforms TCP in almost all cases.

The Guaranteed Bandwidth Protocol (GBW) is an access scheme being proposed for implementation of connection oriented services in DQDB networks. Connection oriented services are expected to handle both constant bit rate (CBR) and variable bit rate (VBR) traffic that have delay and jitter constraints. It has been reported that the GBW protocol can provide guaranteed bandwidth and lower delays compared to the ordinary DQDB protocol. However, the intensity of the jitter introduced by this scheme has not been made clear. This paper compares the jitter results for the GBW scheme to those obtained by a new access method called Modified Guaranteed Bandwidth (MOD_GBW) protocol, which is proposed here. It is shown through simulation that MOD_GBW also provides guaranteed bandwidth and that its delay and jitter characteristics are significantly better than those of the GBW protocol. In the simulation model, the DQDB stations are divided into two groups: 1)Real-Time (RT) stations that generate either CBR or VBR real-time traffic; and 2)Data stations that generate memoryless type of traffic. Data stations operate according to the ordinary DQDB protocol only. The main performance measure adopted here for the real-time traffic is the interdeparture time distribution of consecutive segments from an RT-station. We define the variance of this distribution as jitter. This paper also shows the impact of GBW/MOD_GBW on the performance of the data stations by evaluating their throughput and average bus access delay. Finally, we show that the network performance is weakly related to the number of RT-stations under MOD_GBW, but it depends strongly on the overall loading.

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, an intra-stream and an inter-stream synchronization mechanisms which were previously 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 control of joining an in-progress multicast group. The paper presents experimental results using an interconnected ATM-Ethernet LAN, which is a kind of heterogeneous network. In our experimental system, stored voice and video streams are multicast from a source to plural destinations distributed among distinct networks, and then they are synchronized and output. Furthermore, the paper demonstrates the effectiveness of the mechanism.

This paper focuses on a single BSA (Basic Service Area) in an infrastructure network and studies the performance of the IEEE 802.11 standard MAC protocol by means of simulation. The MAC protocol supports DCF (Distributed Coordination Function) and PCF (Point Coordination Function). The simulation model includes both data transmission with the DCF and H.263 video transmission with the PCF. In the simulation we assume that the channel transmission rate is 2 Mbps and use the system parameters specified in the standard for the DSSS (Direct Sequence Spread Spectrum) physical layer. We evaluate the performance of this protocol in terms of throughput and MPDU (MAC Protocol Data Unit) delay for various values of the CFP (Contention Free Period) repetition interval and the CFP maximum duration. Numerical results show that if the CFP repetition interval is set too long, video MPDU delay becomes very large periodically; therefore, average video MPDU delay deteriorates. We also find that as the CFP maximum duration decreases, the number of video terminals that can be accommodated in the system decreases. Furthermore, how channel transmission errors affect the performance of the protocol is examined. A two-state continuous-time Markov model is used as a burst error model. As a result, we see that for a small number of video terminals, the average video-MPDU-delay performance does not deteriorate drastically for larger values of bit error rate.