This paper presents the development of a sound–specific vibration interface and its evaluation results by playing three commercial games with the interface. The proposed interface complements the pitfalls of existing frequency–based vibration interfaces such as vibrating headsets, mouses, and joysticks. Those interfaces may bring negative user experiences by generating incessant vibrations because they vibrate in response to certain sound frequencies. But the proposed interface which responds to only target sounds can improve user experiences effectively. The hardware and software parts of the interface are described; the structure and the implementation of a wrist pad that delivers vibration are discussed. Furthermore, we explain a sound-matching algorithm that extracts sound characteristics and a GUI-based pattern editor that helps users to design vibration patterns. The results from evaluating the performance show that the success ratio of the sound matching is over 90% at the volume of 20 dB and the delay time is around 400 msec. In the survey about user experiences, the users evaluates that the interface is more than four times effective in improving the reality of game playing than without using the vibration interfaces, and two times than the frequency–based ones.

This paper proposes a scheme for fairness between uplink and downlink in error-prone 802.11 DCF WLANs by differentiating the contention window of AP. While existing schemes consider only collision, the proposed scheme takes into account packet error due to poor channel condition, too. Instead of complex analytical models based on Markov chain processes, a simpler model based on mean value analysis is proposed. It works on 802.11 DCF and so avoids being dependent on TXOP which lacks applicability. A performance evaluation shows that the proposed method can achieve fairness even in error-prone environments without decrease of total throughput when compared with existing schemes.

Existing filtering methods of TCP ACK packets are known to be effective in reducing the required bandwidth, resulting in the improvement of TCP throughput. However, the methods cannot handle the filtering of piggyback ACK packets. Considering that most TCP applications require bidirectional data exchange, the lack of the functionality to deal with the piggyback ACK packets should be addressed. This paper proposes a novel filtering scheme for WiMAX systems that can handle the piggyback ACK packets. The novelty comes from the fact that the proposed method overlaps the processing time of packet merging with the round trip delay of the bandwidth request-and-grant procedure. It is advantageous because it does not require extra time for the merging. The results from an analytical model and simulations show that the required uplink bandwidth is decreased while the downlink throughput is increased.

Multiple-attribute based handoff schemes suffer from instability because of the dynamic nature of attributes and the distribution of handoff procedure over candidate networks, resulting in frequent handoffs that degrade the efficiency of resource management. To alleviate such instability, a service-history based scheme was proposed but it has several improper design decisions, e.g. it considers the history factors too optimistically and employs fixed weights that are likely to distort handoff decisions. This letter proposes to improve handoff performance by considering network state along with the service history. It takes into account the network utilization to avoid the optimistic dependency on the history and adaptively determines the weight to the service history in order to adjust its effect on the handoff decision. Simulation results show that the proposed scheme optimizes the number of handoff and the dropping probability when compared with existing schemes.

In this paper, we propose a novel dynamic bandwidth allocation scheme for the downlink real-time video streaming in the wireless cellular networks. Our scheme is able to maximize the bandwidth utilization, while satisfying the required packet loss probability, a QoS constraint, by dynamically determining the amount of bandwidth to be allocated at each unit time interval by measuring the queue length and the packet loss probability. The simulation results show that, without the need of a priori knowledge about the traffic traces, our scheme is able to achieve the same level of performance as what can be accomplished with the pre-calculated effective bandwidth in terms of the bandwidth utilization and the packet loss rate.

3GPP evolved packet system (EPS) is an all-IP based system that supports various access networks such as LTE, HSPA/HSPA+ and non-3GPP networks. Recently, the support of IP flows with packet level QoS profiles has been added to the requirements of the EPS. This paper proposes an adaptive modulation and coding (AMC) scheme that supports the QoS of such IP flows in the 3G LTE access network of the EPS. Defining the retransmission as a critical factor for QoS, the proposed scheme applies different maximum packet error probability Pmax to each packet when selecting the AMC transmission mode. In determining Pmax, the QoS constraints as well as channel condition are considered, balancing two objectives: the satisfaction of the QoS and the maximization of spectral efficiency. Simulations show that it is able to reduce both delay violation and retransmission, while improving throughput in comparison with an existing scheme.

An active queue management (AQM) scheme is proposed to reduce throughput bias for UDP flows over TCP. It is argued that existing AQM methods partially involve a flow-indifferent factor that does not take into account of bandwidth usage of flows when they determine packet drop, thus resulting in unfairness. The proposed scheme replaces the flow-indifferent part with a flow-wise one by approximating per-flow fair share, which permits the discrimination of unresponsive flows. Since it is a stateless process, it avoids the overhead of tracking the statistics of flows and implementation is simple. A performance evaluation shows that it effectively limits the bandwidth of unresponsive flows to their fair share of bandwidth. In addition, it can also encourage RTT-fairness among TCP flows with different delays.