A number of flexible control mechanisms used in buffer management, congestion control and bandwidth allocation have been proposed to improve the performance of ATM networks by introducing parameters, such as threshold, push-out probability and incremental bandwidth size of a virtual path, which are adjustable by network providers. However, it is difficult to adaptively adjust these parameters, since the traffic in ATM networks is further complicated by accommodating various kinds of services. To overcome the problem, we propose in this paper a control scheme based on the genetic algorithms and the neural estimator. The neural estimator forecasts the future QOS values for each candidate parameter set, and the genetic algorithms select the best one to control the real network. An example of buffer management in an ATM switch is examined in this paper. Simulation results show the effectiveness of the proposed control scheme in adaptively adjusting the parameter set even when the traffic environment and the QOS requirements are dynamically changing.

This paper investigates the performance of an integrated voice/data wireless mobile network where a finite buffer is provided for voice calls since they can endure a tolerable time, or the reneging time, for service. Based on a given humanistic reneging time, we analyze the voice traffic blocking probability. The probability distribution of receiving service within the reneging time is obtained for each buffered voice call and based on this result, an appropriate amount of voice buffer is obtained. To alleviate the impact on data blocking probability caused by the voice buffer and to enhance the efficiency of data service, a dynamic multi-channel allocation scheme with channel de-allocation and guard channels is proposed for data traffic. Compared with the conventional method where the system adopts a single-channel allocation scheme without guard channel for data users, the proposed scheme shows significant improvement in data blocking probability, throughput and the mean service time. Furthermore, a system with an appropriate size of buffer for voice traffic can receive good improvement in voice blocking probability.

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 Public Wireless Local Area Network (PWLAN) is an emerging service for wireless access to the Internet. However, the service coverage of the PWLAN is limited by the deployment of access points (APs) because only those who stay near the AP can access the PWLAN. A feasible way of extending the service coverage of a PWLAN is to deploy mobile ad hoc access networks (MAHANs) so that users who are not in an AP's radio coverage area can send their packets to the AP in a multihop manner. However, in a MAHAN, mobile nodes that intend to access the Internet have to discover routes to the AP first, which may result in considerable bandwidth cost. In this paper, we propose the Appointed BrOadcast (ABO) method to reduce the cost of route discovery in MAHANs. Using the ABO method can achieve this goal on the basis of packet overhearing. Functions that are necessary for network and data link layers to employ the ABO method are also discussed. Simulation results show that using the ABO method can significantly reduce the cost on route discoveries. Due to the widespread use of legacy IEEE 802.11 nodes, the problem of how ABO-enhanced and legacy IEEE 802.11 nodes can coexist in a MAHAN is also discussed.

This paper proposes an analytical model to demonstrate the benefit of data service in wireless networks using dynamic multi-channel scheme with channel de-allocation. The performance of a system providing buffers to voice calls to reduce the raised voice blocking probability caused by data contention is investigated. The effect of the cell dwell time and overlap area with adjacent cells on system performance are studied. All free channels are allocated to data users dynamically. For those data users using more than one channel, channels would be de-allocated for new requests, voice or data. Buffers are provided for voice calls to reduce the voice blocking probability caused by data packets contention. Handoff calls are given priority to be queued in the front of the buffer instead of providing guard channels to reduce their dropping probability. Meanwhile, the reneging time for new calls and the handoff dwell time for handoff calls are considered in our analysis to obtain an appropriate amount of buffer to voice. To compensate the blocking probability in data, guard channels are provided for data traffic. Numerical results show that the dynamic multi-channel scheme with possible de-allocation, compared with the single channel scheme, can enhance data traffic performance significantly in terms of the mean transmission time and blocking probability. A system providing an appropriate amount of buffer to voice traffic and giving priority to queued handoff calls can indeed reduce new call blocking probability and handoff call dropping probability. In addition, the proposed scheme can reduce the incomplete transmission probability of data packets.

In this paper, we analyzed the performance of dynamic resource allocation with channel de-allocation and buffering in cellular networks. Buffers are applied for data traffic to reduce the packet loss probability while channel de-allocation is exploited to reduce the voice blocking probability. The results show that while buffering data traffic can reduce the packet loss probability, it has negative impact on the voice performance even if channel de-allocation is exploited. Although the voice blocking probability can be reduced with large slot capacity, the improvement decreases as the slot capacity increases. On the contrary, the packet loss probability increases as the slot capacity increases. In addition to the mean value analysis, the delay distribution and the 95% delay of data packets are provided.

In this paper, two models are proposed for the simulation of MPEG video sources in ATM networks. The projected autoregressive (PAR) model is based on the autoregressive (AR) model compensated by a projection function. The projection function is capable of adjusting the histogram generated by the AR model so that it better fits the histogram obtained from real data. The state transition (ST) model is developed on die basis of recording the variation of frame size in a video sequence. Each state denotes the size of a frame and the number of state depends on the degree of correlation between frames. Our results show that the histogram generated by the ST model is almost identical with that of the real data and the PAR model performs better in capturing the property of autocorrelation of real data. When compared with other models, both of the two models demonstrate an excellent property of fitting the complex histogram curve, which was not achieved by the AR model, and preserving the correlation characteristics. A heuristic search algorithm is also proposed to make our modelling processes more efficient.

In their paper[1], Sasase and Mori proposed a method of calculating the resequencing delay for a multiple-server queueing system under a threshold-type scheduling. However, it is found that the result of the proposed method does not converge to the corresponding expression in Ref.[2] for the system with two servers. In this correspondence, we show that the proposed method in Ref.[1] is not correct, and verify it by simulation.