In this letter, we propose a new QoS scheduling algorithm, referred to as the explicit traffic aware scheduling algorithm with explicit queue length notification (ETA-EQN), which aims at supporting multimedia services in HCCA of IEEE 802.11e WLAN. The results from intensive simulations with NS-2 verify that ETA-EQN provides much better network performance than the reference scheduler in terms of throughput, delay, and packet loss.

In our previous study, a distortion measure based variable bit rate (DM-VBR) scheme in waveform interpolation (WI) coders was proposed. In this paper, the repetition method is proposed to estimate non-transmitted parameters instead of the extrapolation method. For the further reduction of slowly evolving waveform (SEW) bit rates, the dimensions of the past parameters, which are different from those of the current parameters, are converted to match the dimension of the current ones. Distortions between interpolated sub-frames and original sub-frames are measured for the reduction of the SEW parameters. And the usefulness of several other distortion measures is also investigated instead of the simple log spectral distortion. Experimental results show that the coder adopting the new schemes offers above 41% bit rate reduction with almost unnoticeable output speech degradation.

In this paper, we propose a new variable-bit-rate speech coder based on the waveform interpolation concept. After the coder extracts all parameters, the amounts of distortions between the current and the predicted parameters, which are estimated by extrapolation using the past two parameters, are measured for all parameters. A parameter is not transmitted unless the distortion exceeds the preset threshold. At the decoder side, the non-transmitted parameter is reconstructed by extrapolation with the past two parameters used to synthesize signals. In this way, we can reduce 26% of the total bit rate while maintaining the speech quality degradation below the 0.1 perceptual evaluation of speech quality (PESQ) score.

Packet loss and delay cause degradation in the quality of real-time, interactive applications such as video conferencing. Forward error correction (FEC) schemes have been proposed to make the applications more resilient to packet loss, because the time required to recover the lost packets is shorter than that required to retransmit the lost packets. On the other hand, the codec generally used in real-time applications like MPEG4 has the feature that the sending bit rate and the packet size of the traffic vary significantly according to the motion of an object in a video. If the traditional FEC coding, which is calculated on the basis of a fixed-size block, is applied to such applications, a waste of bandwidth and a delay variation are caused and the quality is degraded. In this paper, we propose suitable FEC schemes for visual communication systems using variable bit-rate (VBR) codec and evaluate the effectiveness of these schemes using our prototype implementation and experimental network.

In this letter, a resource allocation scheme is proposed to efficiently utilize the resource of CDMA systems with respect to the throughput. When we let the throughput be a function of various data rates for multimedia traffics, the scheme is to find the optimum set of data rates for concurrent user groups with which we can achieve the maximum throughput. It is also shown that the optimum data rate set always includes one non-trivial rate while all the others keep the minimum required data rate.

The Quality of Service (QoS) requirements such as delay and cell loss ratio (CLR) are very stringent for video transmission. These constraints are difficult to meet if high network utilization is desired. Dynamic bandwidth allocation in which video traffic prediction can play an important role is thus needed. In this paper, we suggest to predict the variation of I frames instead of the actual size of I frames, and propose an algorithm that can achieve fast convergence and small prediction error, thus imposing QoS and attaining high network utilization. The performance of the scheme is studied using the renegotiated constant bit rate (RCBR) service model. The overall dynamic bandwidth allocation scheme based on our fast convergent algorithm is shown to be promising, and practically feasible for efficient transmission of real time videos.

In this paper, we propose an efficient quality-providing scheme to satisfy delay bound and loss ratio requirements for real-time video applications. To utilize network resources more efficiently while meeting service requirements, the network resources are dynamically allocated to each video connection based on the predicted traffic and currently provided quality of service degree. With the proposed bandwidth allocation method, a fair quality of service support in terms of packet loss ratio and maximum packet transfer delay to each video source can be achieved. To avoid possible quality violation by incoming new video connections, we present a connection admission control based on the provided QoS for existing connections and the measured traffic statistics. Simulation results show that our proposed dynamic method is able to provide accurate quality control.

This paper presents an efficient scheme for access bandwidth control for VBR (Variable Bit Rate) traffic between radio mobile terminals and their base stations in a WATM (Wireless ATM) network. After introducing the wireless ATM system model, we describe a new algorithm that enables dynamic slot allocation under TDMA/TDD (Time Division Multiple Access/Time Division Duplex) Media Access Control, making use of UPC (Usage Parameter Control) parameters and traffic characteristics. We show more efficient bandwidth utilization with our proposed algorithm, compared to other conventional algorithms. Moreover, we reveal that our algorithm improves cell transmission delays.

This paper presents a real-time integrated traffic management (RITM) scheme that can effectively manage variable bit rate (VBR) and available bit rate (ABR) traffics having unpredictable characteristics in asynchronous transfer mode (ATM) networks. An unique feature of this scheme is that it has a special ATM cell control block, which makes it possible to monitor VBR and bursty traffics in real-time so that the delay incurred to measure cell arrival rate is minimized. In addition to its traffic monitor capability, the proposed scheme intends to dynamically reassign the leftover network resources to VBR/ABR connections without any deterioration in quality of service (QoS) of the existing connections. Particularly, we investigate into some functions of traffic management schemes that should be integrated to operate correctly in the emerging ATM networks. Also, we emphasize the merits of the RITM that has been developed to be a simple and efficient traffic management scheme to support the increasing data/Internet services and to get over difficulties in hardware complexities. The capability of managing the incoming ATM traffics in real-time helps determine an optimal acceptable number of user connections for a given network condition. We can use this value as a threshold to prevent the network from being congested and to find out a cost-effective buffer design method. This property makes the scheme extremely attractive as it supports most delay-sensitive and loss-sensitive applications at the same time without changing the existing ATM switching system architecture. The RITM scheme has been verified to reliably monitor incoming traffics and to efficiently manage network resources by computer simulations.

Two main characteristics of VBR MPEG video traffic are the different statistics according to different picture types and the periodic traffic pattern due to GOP structure. Especially, the I-pictures at the beginning of each GOP generate much more traffic than other pictures. Therefore, when several VBR MPEG video sources are superposed, the multiplexing performance can vary according to the variations of their I-picture start times. In this paper, we show how the start time arrangement of the superposed VBR MPEG videos can significantly affect the cell loss ratio characteristics at the multiplexers, by using U-NDPP/D/1/B queueing model. It is also shown that the Lognormal distribution is more suitable for modeling VBR MPEG video traffic than the Normal and Gamma distributions, in the queueing application's view points.

The design of effective traffic and resource management policies is a key issue in the deployment of ATM-satellite systems. This paper proposes a technique of call admission control and dynamic resource management to support ATM traffic classes in satellite environments. The effectiveness of the strategy is assessed by referring to the EuroSkyWay multimedia satellite platform, based on Ka-band payload and on-board processing. The main advantage is the effective exploitation of the satellite bandwidth by means of the statistical multiplexing of traffic sources and the guarantee of QoS provisioning to both real-time and non real-time, constant and variable bit rate sources.

In this paper we present two traffic control approaches, a circuit emulation traffic control (CETC) and an adaptive priority traffic control (APTC) for supporting voice services in ATM networks. Most voice services can be handled as CBR traffic, this causes a lot of wasted bandwidth. Sending voice through VBR (variable bit rate) may be a better alternative, because it allows the network to allocate voice bandwidth on demand. In CETC, the service discipline guarantees the quality of service (QOS) for voice circuits. Through mathematical analysis, we show that CETC features an adequate performance in delay-jitter. Moreover, it is feasible in implementation. We also present an APTC approach which uses a dynamic buffer allocation scheme to adjust the buffer size based on the real traffic need, as well as employs an adaptive priority queuing technique to handle various delay requirements for VBR voice traffic. It provides an adequate QOS for voice circuits in addition to improving the multiplexing gain. Simulation results show that voice traffic get satisfied delay performance using our approaches. It may fulfill the emerging needs of voice service over ATM networks.

This paper investigates the lossless video aggregation and transport, a concept of transmitting a group of video sessions as a bundle over a communication network. We focus on the design of optimal transmission plan with minimum receiver buffer requirement for stored variable bit-rate (VBR) videos over a constant bit-rate (CBR) channel without incurring buffer underflow and overflow. For a single video, an efficient algorithm is proposed to calculate the optimal transmission plan in only O(N) time, a significant improvement of the previous result with time complexity of O(N2log N). For multiple video sessions, we propose an aggregation scheme to calculate the changes of optimal transmission plans at the joints or separations, where a session bundled in the aggregated video stream begins or stops. The experimental results show that our algorithms stand out in terms of simplicity and efficiency.

The proposed GCRA (Generic Cell Rate Algorithm) traffic shaper consists of a regulator and a scheduler. It can shape multiple incoming VBR (Variable Bit Rate) cell streams simultaneously to be strictly conforming according to the GCRA algorithm when the cells depart for the ATM output link. The impact of cell emission conflicts is considered and resolved by using an EDD (Earliest-Due-Date) scheduler and a feedback signal from the scheduler to the regulator. The call admission control condition and the cell delay bound are derived. Simulation results demonstrate that the output cell streams of the proposed GCRA traffic shaper do not contain any non-conforming cells and the scheduler queue size is significantly reduced. Meanwhile, the delay performance is almost not affected by the use of the feedback mechanism.

This paper presents the system issues involved with the transmission of pre-encoded VBR MPEG video using CBR service. Conventional wisdom suggests that lossless delivery of VBR video using CBR service requires bandwidth to be reserved at the peak rate resulting in low bandwidth utilization. We calculate the minimum rate at which bandwidth must be reserved on a network in order to provide continuous playback of an MPEG encoded video bitstream. Simulation results using the frame size traces from several pre-encoded MPEG bitstreams and several buffer sizes demonstrate that this minimum reservation rateis much lower than the peak rate when a relatively small playback buffer size is used, resulting in much higher bandwidth utilization. Procedures for performing connection setup and lossless realtime video playback between the video server and the client are outlined. Methods for incorporating VCR-like features such as pauseandfast forward/reversefor Video-on-Demand (VoD) applications are presented.