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.

The proportional delay differentiation (PDD) model provides consistent packet delay differentiation between classes of service. Currently, the present schedulers performing the PDD model cannot achieve desired delay proportion observed in short timescales under light/moderate load. Thus, we propose a Non-Work-Conserving (NWC) scheduler, which utilizes the pseudo-waiting time for an empty queue and forces each class to compare its priority with those of all other classes. Simulation results reveal that NWC outperforms all current schedulers in achieving the PDD model. However, NWC suspends the server from transmitting packets immediately if an empty class has the maximum priority, resulting in an idle server. Therefore, we further propose two approaches, which will serve a best-effort class during this idle time. Compared with other schedulers, the proposed approaches can provide more predictable and controllable delay proportion, accompanied with satisfactory throughput and average queuing delay.

In this letter, we derive an efficient audio/video synchronization method for video telephony. For synchronization, this method does not require any further RTCP packet processing except for the first one. The derived decision rule is far more compact than the conventional method. This decision rule is incorporated in an actual video telephony system adopting Texas Instruments (TI) OMAP 1510 processor and Qualcomm MSM 5500. The computational requirement was compared with the conventional method and through simulations the superiority of the proposed method is proved.

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 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 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 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.

This paper proposes a scheme for joint synchronization between stored media with interactive control and live media in multicast communications. We deal with visual search control, such as fast-forward and fast-reverse, as interactive control. The proposed scheme enables visual search by enhancing the virtual-time rendering (VTR) media synchronization algorithm, which the authors previously proposed, and adjusts the timing of changing the visual search mode among destinations by carrying out group synchronization control. We also demonstrate the effectiveness of the scheme by experiment.

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 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 Δ-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 an object-oriented model to handle the temporal relationship for all of the multimedia objects at the presentation platform. Synchronization of the composite media objects is achieved by ensuring that all objects presented in the upcoming "manageable" period must be ready for execution. To this end, the nature of overlays is first investigated for various types of objects. Critical overlaps which are crucial in synchronization are also defined. The objective of synchronization is to ensure that the media objects can be initiated precisely at the critical point of the corresponding critical overlap. The concept of manageable presentation interval is introduced and the irreducible media group is defined. The resource scheduling of each presentation group for media object pre-fetch time versus buffer occupancy is also examined. Accordingly, a new model called group cascade object composition Petri-net (GCOCPN) is proposed and an algorithm to implement this temporal synchronization scheme is presented.

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.

Synchronization and continuity are essential for multimedia presentation, but because network resources and available bandwidth are both limited, synchronization quality and continuity quality have to be traded off in response to the fluctuating network conditions. This paper therefore introduces an algorithm for intramedia synchronization with adaptive quality of service (QoS) control handled at different layers of multimedia streams. The work described here is an extension of our earlier proposal of a synchronization algorithm by delay compensation protocol with two resynchronization mechanisms: retrieval offset adjustment and data unit skipping. That algorithm has been extended by the introduction of QoS control mechanisms in the QoS plane of a distributed control platform. The extended approach results not only in better synchronization and continuity, but also integrates the QoS adjustment into the existing architecture. Unexpected QoS variations are coped with by an adaptive QoS control designed to maintain the desired application qualities within the fluctuating environment. Simulations implemented on a UDP/IP network have verified the effectiveness of the proposed scheme.

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 presents results of subjective assessment of the media synchronization quality in the virtual-time rendering (VTR) media synchronization algorithm. For the assessment, stored voice and video streams were transmitted as two separate transport streams from a source to a destination on various traffic conditions in an experimental system. At the destination, they were output after synchronization control. We subjectively assessed the quality of media synchronization in a systematic way. This paper examines the effects of the difference between methods of recovery from asynchrony on the media synchronization quality. The paper also clarifies the relationships between the subjective and objective performance measures. Furthermore, it examines the effect of the difference in scene between media streams and that of the modification of the target output time on the media synchronization quality.

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.

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 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 proposes a bitstream scaling technique for MPEG video for the purpose of media synchronizations. The proposed scaling technique can reduce the frame rate as well as the bit rate of an MPEG data sequence to fit them to the values specified by a synchronization system. The advantage of the proposed technique over existing scaling methods is that it is considering not only the performance of synchronization but also the picture quality of the resulting sequences. To further improve the quality of sequences scaled by the proposed method, this paper also proposes an MPEG encoding technique which sets some of the parameters suitable for the scaling. An experiment using these techniques in an actual media synchronization system has illustrated the usefulness of the proposed approach.

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.