In this paper, we extend the derivation of the iterative soft-output Viterbi algorithm (SOVA) for partial response (PR) channels, and modify its decoding process such that it works consistently for arbitrary high code rates, e.g., rate 64/65. We show that the modified SOVA always outperforms the conventional SOVA that appears in the literature with a significant difference for high code rates. It also offers a significant cut down in the trace-back computations. We further examine its performance for parallel and serial concatenated codes on a precoded Class IC partial response (PR4) channel. Code rates of the form $\frac{k_0}{k_0+1}$($k_0$ = 4, 8, and 64) are considered. Our simulations indicate that the loss suffered by the modified SOVA, relative to the APP algorithm, is consistent for all code rates and is at most 1.2 dB for parallel concatenations and at most 1.6 dB for serial concatenations at $P_b$ = $10^{-5}$.

In this paper, a decision feedback recurrent neural network equalizer and a modified real time recurrent learning algorithm are proposed, and an adaptive adjusting of the learning step is also brought forward. Then, a complex case is considered. A decision feedback complex recurrent neural network equalizer and a modified complex real time recurrent learning algorithm are proposed. Moreover, weights of decision feedback recurrent neural network equalizer under burst-interference conditions are analyzed, and two anti-burst-interference algorithms to prevent equalizer from out of working are presented, which are applied to both real and complex cases. The performance of the recurrent neural network equalizer is analyzed based on numerical results.

A new code, LS code, was proposed for the IMT-2000 CDMA system. The code has special properties during a specific time interval: 1) perfect autocorrelation, and 2) perfect crosscorrelation. Perfect autocorrelation means maximum autocorrelation for zero time-offset and zero autocorrelation for all other times during a specific time interval. Moreover, perfect crosscorrelation means that the crosscorrelation has zero value during a certain time, you can remove all the MAI within that time. However, the detailed properties of LS code and its exact generation method have been previously unknown. Therefore, we investigate the LS code in regards to its exact generating method, properties, and performances in this paper.

4G mobile communication system requires the throughput of 10-100Mbps. Adaptive modulated OFDM system is promising technique for increasing the throughput. In the pilot symbol assisted high-speed packet transmission system, the data symbol duration is generally considered to be small compared to the coherence time. However, OFDM symbol duration is longer than the symbol duration of a single carrier system, so that the packet duration of the pilot symbol assisted high speed packet transmission system is long. In this case, the change of channel conditions is too fast to be accurately estimated by channel estimator at the receiver in high Doppler frequency, so that many errors occur during demodulation, especially with the data symbols at the end of each packet. In this paper, we consider the BER at various instantaneous $E_b/N_o$ that includes the demodulation errors in high Doppler frequency. When the coherence time is ten times longer than the duration of a single packet, the channel can be closely approximated as an AWGN channel. Otherwise, the approximation breaks down and the above-mentioned errors that occur during demodulation must be taken into consideration. In this paper, we propose the pilot symbol assisted high speed packet transmission system based on adaptive OFDM using a novel lookup table to consider the demodulated errors and evaluate the throughput performance.

In this paper a two-stage maximum-likelihood (ML) detection structure for group detection in DS/CDMA systems is presented. The first stage of the receiver is a linear filter, aimed at suppressing the effect of the unwanted (i.e., out-of-grout) users' signals, while the second stage is a non-linear block, implementing a ML detection rule on the set of desired users signals. As to the linear stage, we consider both the decorrelating and the minimum mean square error approaches. Interestingly, the proposed detection structure turns out to be a generalization of Varanasi's group detector, to which it reduces when the system is synchronous, the signatures are linerly independent and the first stage of the receiver is a decorrelator. The issue of blind adaptive receiver implementation is also considered, and implementations of the proposed receiver based on the LMS algorithm, the RLS algorithm and subspace-tracking algorithms are presented. These adaptive receivers do not rely on any knowledge on the out-of group users' signals, and are thus particularly suited for rejection of out-of-cell interference in the base station. Simulation results confirm that the proposed structure achieves very satisfactory performance in comparison with previously derived receivers, as well as that the proposed blind adaptive algorithms achieve satisfactory performance.

Due to insufficient available bandwidth resources and the continuously growing demand for cellular communication services, the channel assignment problem has become increasingly important. To trace the optimal assignment, several heuristic strategies have been proposed. So far, most of them focus on the small-scale systems containing no more than 25 cells and they use an anachronistic cost model, which does not satisfy the requirements ity. Solving the small-scale channel assignment problems could not be applied into existing large scale cellular networks' practice. This article proposes a decomposition approach to solve the fixed channel assignment problem (FCAP) for large-scale cellular networks through partitioning the whole cellular network into several smaller sub-networks and then designing a sequential branch-and-bound algorithm that is made to solve the FCAP for them sequentially. The key issue of partition is to minimize the dependences of the sub-networks so that the proposed heuristics for solving smaller problems will suffer fewer constraints in searching for better assignments. The proposed algorithms perform well based on experimental results and they were applied to the Taiwan Cellular Cooperation (TCC) in ChungLi city to find better assignments for its network.

Mobile IP has several inefficiencies, and was not originally designed for situations where both peers are highly mobile. We present a mobility management solution that retains compatibility with existing Internet protocols, whilst increasing the efficiency of communications between two GPRS mobile hosts. Our proposal eradicates triangle routing and minimizes handover latency. We show by numerical analysis that the routing optimization improves the performance of TCP controlled data flows, reducing buffering requirements and minimizing the recovery time after a handover occurs.

Internet roundtrip delay/time (RTT) prediction plays an important role in detecting packet losses in reliable transport protocols for traditional web applications and determining proper transmission rates in many rate-based TCP-friendly protocols for Internet-based real-time applications. The widely adopted autoregressive and moving average (ARMA) model with fixed-parameters is shown to be insufficient for all scenarios due to its intrinsic limitation that it filters out all high-frequency components of RTT dynamics. In this paper, we introduce a novel parameter-varying RTT model for Internet roundtrip time prediction based on the information theory and the maximum entropy principle (MEP). Since the coefficients of the proposed RTT model are updated dynamically, the model is adaptive and it tracks RTT dynamics rapidly. The results of our experiments show that the MEP algorithm works better than the ARMA method in both RTT prediction and RTO estimation.

The controlled load service defined within the IETF's Integrated Services architecture for quality-of-service (QoS) in the Internet requires source nodes to regulate their traffic while the network, in combination with an admission control strategy, provides a guarantee of performance equivalent to that achieved in a lightly loaded network. Packets sent in violation of the traffic contract are marked so that the network may assign them a lower priority in the use of bandwidth and buffer resources. In this paper, we define the requirements of a scheduler serving packets belonging to the controlled load service and present a novel scheduler that exactly achieves these requirements. In this set of requirements, besides efficiency and throughput, we include an additional important requirement to bound the additional delay of unmarked packets caused due to the transmission of marked packets while dropping as few marked packets as possible. Without such a bound, unmarked packets that are in compliance with the traffic contract are not likely to experience delays consistent with that in al lightly loaded network. For any given desired bound ${\alpha}$ on this additional delay, we present the CL(${\alpha}$) scheduler which achieves the bound while also achieving a per-packet work complexity of O(1) with respect to the number of flows. We provide an analytical proof of these properties of the CL(${\alpha}$) scheduler, and we also verify this with simulation using real traces of video traffic. The scheduler presented here may be readily adapted for use in scheduling flows with multi-level priorities such as in some real-time video streams, as well as in other emerging service models of the Internet that mark packets to identify drop precedences.

In this paper, an efficient threshold-based filtering (TF) buffer management scheme is proposed. The TF is capable of minimizing the overall loss performance and improving the fairness of buffer usage in a shared buffer packet switch. The TF consists of two mechanisms. One mechanism is to classify the output ports as sctive or inactive by comparing their queue lengths with a dedicated buffer allocation factor. The other mechanism is to filter the arrival packets of inactive output ports when the total queue length exceeds a threshold value. A theoretical queuing model of TF is formulated and resolved for the overall packet loss probability. Computer simulations are used to compare the overall loss performance of TF, dynamic threshold (DT), static threshold (ST) and pushout (PO). We find that TF scheme is more robust against dynamic traffic variations than DT and ST. Also, although the over-all loss performance between TF and PO are close to each other, the implementation of TF is much simpler than the PO.