This paper assesses application-level QoS and Quality of Experience (QoE) in the case where audio and video streams are transferred with the enhanced distributed channel access (EDCA) of the IEEE 802.11e MAC. In EDCA, a station can transmit multiple MAC frames during a transmission opportunity (TXOP); this is referred to as TXOP-bursting. By simulation, we first compare application-level QoS with the TXOP-bursting scheme and that without the scheme for various distances between access point (AP) and stations. In this paper, we suppose that the bit error rate (BER) becomes larger as the distance increases. Numerical results show that TXOP-bursting can improve many metrics of video quality such as average media unit (MU) delay, MU loss ratio, and media synchronization quality, particularly when the AP sends audio and video streams to stations in the downlink direction. We then examine the effect of TXOPLimit on the video quality. Simulation results show that the video quality can be degraded if the value of TXOPLimit is too small. Furthermore, we assess QoE by the method of successive categories, which is a psychometric method. Numerical results show that TXOP-bursting can also improve the QoE. We also perform QoS mapping between application-level and user-level with principal component analysis and multiple regression analysis.

This paper performs application-level QoS and user-level QoS assessment of audio-video streaming in cross-layer designed wireless ad hoc networks. In order to achieve high QoS at the user-level, we employ link quality-based routing in the network layer and media synchronization control in the application layer. We adopt three link quality-based routing protocols: OLSR-SS (Signal Strength), AODV-SS, and LQHR (Link Quality-Based Hybrid Routing). OLSR-SS is a proactive routing protocol, while AODV-SS is a reactive one. LQHR is a hybrid protocol, which is a combination of proactive and reactive routing protocols. For application-level QoS assessment, we performed computer simulation with ns-2 where an IEEE 802.11b mesh topology network with 24 nodes was assumed. We also assessed user-level QoS by a subjective experiment with 30 assessors. From the assessment results, we find AODV-SS the best for networks with long inter-node distances, while LQHR outperforms AODV-SS for short inter-node distances. In addition, we also examine characteristics of the three schemes with respect to the application-level QoS in random topology networks.

This paper studies the application of a media synchronization mechanism to the interleaved transmission of video and audio specified by the H.223 Annex in PHS. The media synchronization problem due to network delay jitters in the interleaved transmission has not been discussed in either the Annex or any related standards. The slide control scheme, which has been proposed by the authors, is applied to live media. We also propose a QOS control scheme to control both quality of the media synchronization and that of the transmission delay. Through simulation we confirm the effectiveness of the slide control scheme and the QOS control scheme in the interleaved transmission.

In this paper, we formulate and solve a discrete-time queueing problem that has two potential applications: ATM multiplexers and DQDB networks. We first consider the modeling of an ATM multiplexer. The object of the analysis is a periodic traffic stream (CBR traffic), which is one of the inputs to the multiplexer. As in previous works of the subject, we consider a memoryless background traffic input. Here, in addition to this background traffic, we take into account the influence of a high-priority traffic, which is time-correlated and requires expedited service. We analyze the influence of these two types of traffic on the statistics of the interdeparture time (jitter process) and the delay of the periodic traffic stream. We obtain their distributions in a form of z-transforms, and from these we derive closed form expressions for the average delay and the variance of the interdeparture time. Our results show that the delay and jitter are very sensitive to the burstiness of the high priority traffic arrival process. We next apply our analytical modeling to a DQDB network when some of its stations are driven by CBR sources. We can obtain interesting results concerning the influence of the physical location of a DQDB station on the jitter.

This paper presents an application-level QoS comparison of three inter-destination synchronization 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 for live audio and video streaming over the Internet/intranets. We compare the application-level QoS of these schemes by simulation with the Tiers model, which is a sophisticated network topology model and reflects hierarchical structure of the Internet. The comparison clarifies their features and finds the best scheme in the environment. The simulation result shows that the distributed control scheme provides the highest quality of inter-destination synchronization among the three schemes in heavily loaded networks, while in lightly loaded networks the other schemes can have almost the same quality as that of the distributed control scheme.

This paper presents a performance comparison between the single-stream and the multi-stream approaches to 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. Making the performance measurement of the six schemes, we compare them to identify and evaluate advantages and disadvantages of each approach. We then show that the performance difference between the two approaches is small and that the dynamic resolution control improves the synchronization quality.

This paper proposes the MultiPath streaming scheme with Media Synchronization control (MPMS) for audio-video transmission in wireless ad hoc networks. In many audio-video streaming applications, media compensate each other from a perceptual point of view. On the basis of this property, we treat the two streams as separate transport streams, and then the source transmits them into two different routes if multiple routes to the destination are available. The multipath transmission disturbs the temporal structure of the streams; in MPMS, the disturbance is remedied by media synchronization control. In order to implement MPMS in this paper, we enhance the existing Dynamic Source Routing (DSR) protocol. We compare the application-level QoS of MPMS and three other schemes for audio-video transmission by simulation with ns-2. In the simulation, we also assess the influence of the multipath transmission on other traffic. The simulation result shows that MPMS is effective in achieving high QoS at the application-level.

This paper proposes a scheme for synchronization of stored video and audio streams in 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.

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.