An adaptive rate controller (ARC) based on an adaptive neural fuzzy inference system (ANFIS) is designed to autonomously adjust the data rate of a mobile heterogeneous network to adapt to the changing traffic load and the user speed for multimedia call services. The effect of user speed on the handoff rate is considered. Through simulations, it has been demonstrated that the ANFIS-ARC is able to maintain new call blocking probability and handoff failure probability of the mobile heterogeneous network below a prescribed low level over different user speeds and new call origination rates while optimizing the average throughput. It has also been shown that the mobile cognitive wireless network with the proposed CS-ANFIS-ARC protocol can support more traffic load than neural fuzzy call-admission and rate controller (NFCRC) protocol.

In this paper, we study the power control for the adaptive rate system that adapts the data rate to the channel and interference condition by adjusting the SINR requirement. The power control technique aims to decrease the interference to obtain the satisfactory throughput. We first find the optimal CINR that allows as many users as possible to transmit their data with the satisfied throughput requirements. Then we use the optimal CINR as the target CINR in the proposed distributed power control algorithm. During the operation of this algorithm, we let each user adjust the transmit power for several iterations to obtain the satisfactory CINR, which in turn leads to the satisfactory throughput. We show that the user can meet its throughput requirement by increasing the number of iterations, and once the user meets its throughput requirement, this condition will hold from then on. Furthermore, it is proven that the throughput can converge to the throughput requirement for each user.

An adaptive rate communication system based on the N-MSK modulation technique is described. Two examples of the system using a 2-MSK adaptive modulation scheme and a 4-MSK adaptive modulation scheme are presented and analysed in slow fading channel. The channel attenuation obeys either Rayleigh or lognormal distribution. The proposed adaptive rate communication system is able to track slow variations of channel attenuation and the average system throughput is therefore increased at a given BER.

As the current Internet becomes popular in information access, demands for real-time display and playback of continuous media are ever increasing. The applications include real-time audio/video clips embedded in WWW, electronic commerce, and video-on-demand. In this paper, we present a new control protocol R3CP for real-time applications that transmit stored MPEG video stream over a lossy and best-effort based network environment like the Internet. Several control mechanisms are used: a) packet framing based on the meta data; b) adaptive queue-length based rate control scheme; c) data preloading; and d) look-ahead pre-retransmission for lost packet recovery. Different from many adaptive rate control schemes proposed in the past, the proposed flow control is to ensure continuous, periodic playback of video frames by keeping the receiver buffer queue length at a target value to minimize the probability that player finds an empty buffer. Contrary to the widespread belief that "Retransmission of lost packets is unnecessary for real-time applications," we show the effective use of combining look-ahead pre-retransmission control with proper data preloading and adaptive rate control scheme to improve the real-time playback performance. The performance of the proposed protocol is studied via simulation using actual video traces and actual delay traces collected from the Internet. The simulation results show that R3CP can significantly improve frame playback performance especially for transmission paths with poor packet delivery condition.

A new adaptive rate control with congestion prediction is developed that is highly robust against long propagation delays. It minimizes the network performance degradation caused by the delay based on prediction by extrapolating past data and correction using new notification. The simulation results show that our proposed control maintains high throughput and a smaller buffer even in long propagation delay networks, like ATM-WAN.

This paper presents an adaptive rate error control scheme for digital communication over time-varying channels. The cyclic code with majority-logic decoding is used in a cascaded way as an inner code to create a simple and powerful hybrid-ARQ error control scheme. Inner code is used only for error correction and the outer code is used for both error correction and error detection. When an error is detected, retransmission is required. The unsuccessful packets are not discarded as with conventional schemes, but are combined with their retransmitted copies. Approximations for the throughput efficiency and the undetectable error probability are given. A high reliability coupled with a simple high-speed implementation makes it suitable for high data rate error control over both stationary and nonstationary channels. Adaptive error control scheme becomes the best solution for time-varying channels when the optimum code is selected according to the actual channel conditions to enhance the system performance. The main feature of this system is that the basic structure of the encoder and decoder need not be modified while the error-correction capability of the code increases. Results of a comparative analysis show that the proposed scheme outperforms other similar ARQ protocols.