Delay jitter degrades the quality of delay-sensitive live media streaming. We investigate the use of multipath transmission with two paths to reduce delay jitter and, in this paper, propose a nearly equal delay path set configuration (NEED-PC) scheme that further improves the performance of the multipath delay jitter reduction method for delay-sensitive live media streaming. The NEED-PC scheme configures a pair of a maximally node-disjoint paths that have nearly equal path delays and satisfy a given delay constraint. The results of our simulation experiments show that path sets configured by the NEED-PC scheme exhibit better delay jitter reduction characteristics than a conventional scheme that chooses the shortest path as the primary path. We evaluate the performance of path sets configured by the NEED-PC scheme and find that the NEED-PC scheme reduces delay jitter when it is applied to a multipath delay jitter reduction method. We also investigate the trade-off between reduced delay jitter and the increased traffic load incurred by applying multipath transmission to more flows. The results show that the NEED-PC scheme is practically effective even if the amount of additional redundant traffic caused by using multipath transmission is taken into account.

The wireless streaming media communications are fragile to the delay jitter because the conditions and requirements vary frequently with the users' mobility. Buffering is a typical way to reduce the delay jitter of media packets before the playback, however, it will incur a longer end-to-end delay. Our motivation in this paper is to improve the balance between the elimination of delay jitter and the decrease of end-to-end delay. We propose a novel adaptive playback buffer (APB) based on the probing scheme. By utilizing the probing scheme, the instantaneous network situations are collected, and then the delay margin and the delay jitter margin are employed to calculate the step length (sl) which is used to adjust the playback buffer in each time. The adaptive adjustment to the playback buffer in APB enables the continuous and real-time representation of streaming media at the receiver. Unlike the previous studies, the novelty and contributions of the paper are: a) Accuracy: by employing the instantaneous network information, the adjustment to the playback buffer correctly reflects the current network situations and therefore achieves the improved balance between the elimination of delay jitter and the decrease of end-to-end delay; Hence, APB adjustment is accurate in terms of improving such balance; b) Efficiency: by utilizing the simple probing scheme, APB achieves the current network situations without the complex mathematic predictions, which enables the adjustment to be more timely and efficient. Performance data obtained through extensive simulations show that our APB is effective to reduce both delay jitter and playback buffer delay.

We study the statistical multiplexing performance of self-similar traffic. We consider that input streams have different QoS (Quality of Service) requirements such as loss and delay jitter. By applying the FBM (fractal Brownian motion) model, we present methods of estimating the effective bandwidth of aggregated traffic. We performed simulations to evaluate the QoS performances and the bandwidths required to satisfy them. The comparison between the estimation and the simulation confirms that the estimation could give rough data of the effective bandwidth. Finally, we analyze the bandwidth gain with priority multiplexing against non-prioritized multiplexing and suggest how to get better performance with the right configuration of QoS parameters.

For several years, more and more people are joining the Internet and various kind of packets (so called transaction-, block-, and stream-types) have been transmitted in the same network, so that poor network conditions cause loss of the stream-type data packets, such as voices, which request smaller transmission delay time than others. We consider a switching node (router) in a network as an N-series M/G/1-type queueing model and have mainly evaluated the fluctuation of packet delay time and end-to-end delay time, using the two moments matching method with initial value, then define the delay jitter D of a network which consists of jointed N switching nodes. It is clarified that this network is not suitable for voice packets transmission media without measures.

Rate monotonic traffic scheduling algorithm had been shown to be the optimal static priority assignment scheme. The system model studied in can be considered (although not realistic) as a preemptive multiplexer which accepts constant bit rate connections that generate packets periodically. The multiplexer adopts a service discipline such that a lower priority packet can be preempted at any stage by a higher priority one without any loss. The constraint is that every packet has to complete its service before the arrival of its succeeding packet generated by the same connection. In this paper, we study the schedulability problem of rate monotonic schedulers for a fixed-length packet switched network such as the ATM network. A necessary and sufficient condition for a set of m constant bit rate connections to be rate monotonic schedulable is first derived and then utilized to design fast admission control algorithms. One of our algorithms computes in advance the minimum period of a connection which can be accepted given a set of existing connections.

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.

A new transfer mode and a switching architecture which can support loss free and no delay jitter service class with shorter switching delay compared with "stop and go queueing scheme" is proposed. This scheme combines ATM scheme with hierarchical STM framing concept.