In this paper, we consider a mobile, multimedia, multihop (M3) ad hoc network. Key characteristics of this system are the mobility of users, energy constraints, and the need to operate without a fixed (wired or wireless) infrastructure. In this environment, with the advent of multimedia communications, the use of the cluster architecture has been revisited to support the resource reservation and Quality-of-Service routing. We proposed an access-based clustering protocol (ABCP) that allows the network to self-organize into a cluster architecture. Three advantages are claimed by ABCP. First, by the access-based criterion, it minimizes the overhead on cluster formation so that the protocol has short execution time and good scalability. Second, ABCP unifies the algorithms for cluster initialization and maintenance, i.e., the same set of clustering functions are used by a node regardless of whether it just becomes active or is in leaving its current cluster. Third, simulation results demonstrate that the cluster structure behaves stably amid topology changes compared with techniques previously proposed. Together with the access-based criterion, a multiple access scheme is also proposed for the broadcast of control messages.

Differentiated services (Diffserv) model is one of the possible solutions for providing quality of service (QoS) on the Internet. Most existing approaches assume that the packet loss is an indication of network congestion and thus reduce the sending rates of sources. For wireless networks, the assumption is not proper since packet losses may be caused by other reasons, such as fading and interference of the signal. Therefore, these approaches do not work well in wireless networks. In this paper, we propose an approach which is able to provide service differentiation in wireless environments. In our approach, the rate share of a connection is determined by the associated weight. By keeping a proper amount of extra data in the network, the proposed approach can achieve weighted proportional fairness, which can provide selective QoS without any particular support from the network. We use the ns simulator to evaluate our approach. Simulation results show the validity of the proposed approach.

An appropriate call admission control in the next generation wireless networks is expected to make efficient use of scarce wireless resource and improve quality-of-service, while supporting multimedia services. On one hand, blocking handoff calls is normally more annoying than blocking new calls. On the other hand, blocking new calls reduces resource utilization. More importantly, handoff call arrival rate depends strongly on call holding time. A novel Call-Holding-Time-Based Random Early Blocking (CHTREB) scheme is proposed to achieve the aforesaid goals in a two-tier cellular voice/data network. With CHTREB, new calls are accepted according to some acceptance probability taking into account the call hloding time difference between voice and data calls. An iterative algorithm is developed to calculate performance measures of interest, i.e., new call blocking probability and forced termination probability. First, simulation results are shown to verify analytical results. Then, numerical results are presented to show the robustness of CHTREB. It is found that CHTREB outperforms TR and CHTREB-FAP under both stationary and nonstationary traffic scenarios. Last but not least, the studied 2-tier system is compared with 1-tier counterpart. It is shown that 2-tier system performs better in terms of average number of handoffs per data call.

We describe a control scheme for wireless-link layers and the TCP/IP layers in which wireless link states, such as signal strength and transmission rate, are transparent to the upper network layers and govern the behavior of the upper layers. Monitoring and notifying functions for wireless link states are incorporated into mobile hosts, and prefetching functions for the mobility agent lists are added to the present MobileIP functionalities of the network layers. In addition, we give the mobile hosts functions for controlling the TCP advertised windows in the transport layers and give the base-stations buffering functions to deal with the variable signal strength of the wireless links. A simulation (using ns-2) of this control scheme shows that mobile agent can be switched at the network layers soon after base-station hand-over and that errors in packet routing, packet loss and communication throughput reduction can be avoided. Moreover, communications can be interrupted without having to shrink the size of the congestion windows of the TCP senders, which improves overall throughput.

Due to the lack of support in mobility and quality of service, today's IP-based networks have some inherent limitations for delivering multimedia services in a mobile environment. In the past few years, these issues have been addressed in the research community and the resulting techniques are being standardised. However, these developments have been done in isolation and become incompatible with each other. In addition to these technical issues, the future infrastructure for charging and accounting mobile multimedia services is expected to be increasingly complicated. In this paper we present a home-proxy based framework which can facilitate the integration of mobility support and QoS management. Furthermore, it enables centralised accounting, which simplifies the cost recovery processes of roaming services. To prove the viability of our design, we have built and tested this framework on a DiffServ wide-area backbone using an MP3 streaming application.

The IETF Mobile IPv6 enables any IPv6 node to both cache the Care-of Address associated with a mobile node's home address, and to directly send packets addressed to a mobile node at the Care-of Address using the IPv6 routing header. Support for optimizing the route is built in as a fundamental part of the protocol. Several hierarchical schemes have been presented recently on top of the Mobile IPv6. These schemes separate micro-mobility from macro-mobility and exploit a mobile node's locality. They can reduce the number of signaling messages sent to a home network and improve hand-off performance. However, existing hierarchical schemes do not achieve route optimization. When external correspondent nodes send packets to a mobile node, these packets are intercepted by an intermediate mobility agent encapsulated and routed to the mobile node. In this paper, we propose a new hierarchical scheme that enables any correspondent node to cache two Care-of Addresses; the mobile node's temporary address and the intermediate mobility agent's address. Also, we introduce two lifetimes managing the two Care-of Addresses. Until the lifetime associated with the mobile node's temporary address expires, a correspondent node can send packets directly to the mobile node. If the lifetime expires but the lifetime associated with the intermediate mobility agent's address has not expired, the correspondent node sends packets to the intermediate mobility agent. This proposal can reduce delay in packet delivery and optimize routing. Furthermore, based on the mobility of a mobile node, we introduce more reduced frequency of binding update and longer period of the lifetimes than the existing hierarchical schemes. Therefore, our proposal can reduce the binding update bandwidth as well as the packet delivery bandwidth lower than those of the IETF IPv6 and the existing hierarchical schemes.

High-speed digital data transmission services with mobile equipment are becoming available. Though the visual signal is one of the expected media to be used with such transmission capabilities, the bandwidth of visual signal is much broader than the provided transmission bandwidth in general. Therefore efficient video encoding algorithms have to be introduced. The ITU-T Recommendation H.263 and ISO/IEC MPEG-4 are very powerful encoding algorithms for a wide range of video sequences. But a large amount of bits are generated in highly active scenes to encode them using such conventional methods. This results in frame skipping and degradation of decoded picture quality. In order to keep these degradations as low as possible, we proposed a Dynamic Resolution Conversion (DRC) method of the prediction error. In the method, a reduced resolution encoding is carried out when the input scene is highly active. Simulation results show that the proposed scheme can improve both coding frame rate and picture quality in a highly active scene. We also present in this paper that some analysis for the performance of the DRC method under the error prone environment that is inevitable with mobile communications.

Error concealment is an essential part of reliable video communication systems because transmission errors are inevitable even when the coded bitstream is highly protected. The problem of temporal EC can be factored into two parts regarding candidate motion vectors (MVs) employed and the matching criterion to evaluate the fitness of each candidate MV. In order to obtain more faithful EC results, this paper proposes a novel iterative EC algorithm, in which an efficient way to provide candidate MVs and a new fitness measure are presented. The proposed approach for candidate MVs systematically utilizes all the available neighboring MVs by exploiting a well-known spatiotemporal correlation of block MVs. Also, in order to remove the dependency of a damaged block's quality of concealment on the already concealed adjacent blocks, we develope a new matching criterion. The objective of the proposed fitness measure is to minimize the total boundary matching errors induced by the whole corrupted blocks. Simulations performed using an H.263 codec demonstrate a significant improvement on the subjective and objective concealed video qualities, especially when the corrupted area is wider than a single row of coding blocks.

We evaluate the performance of indoor infrared wireless systems using on-off keying code division multiple access (OOK-CDMA) with decision-feedback equalizer (DFE) on diffuse channels. To estimate the impulse response, we use the training sequence that alternates '1' and '0. ' We show that the OOK-CDMA with DFE and the training sequence can achieve better performance than the OOK-CDMA without DFE. We also show that the OOK-CDMA with DFE and the training sequence can achieve almost the same performance as the OOK-CDMA with DFE and the known impulse response.

In this paper, a medium access control protocol is proposed for the integrated voice and data services in the local CDMA communication systems. Based on WB-TD/CDMA, uplink channels for the proposed protocol are composed of time slots with multiple spreading codes for each slot. During a talkspurt, a voice terminal transmits its entire packets over a reserved code. On the other hand, a data terminal transmits its packet after sensing the spreading code status. The base station broadcasts the status of spreading codes for each slot. In this protocol, voice packets never contend with data packets. The numerical results show that this protocol increases the system capacity for voice service by applying the reservation scheme. Despite the low access priority of data terminal, the data traffic performance can be increased in proportion to the number of spreading codes.

In this paper, we first investigate the problems of the existing branch point algorithms for available bit rate (ABR) multicast connections in ATM networks, and then propose various solutions for resolving those problems. By combining these solutions, we also propose a new efficient and scalable branch point algorithm. In the proposed algorithm, each branch point stores the feedback information on a per-branch basis for each virtual connection and only passes BRM cells returning from the farthest destination. Simulation results show that the proposed algorithm can provide a good fairness, a higher efficiency and an excellent scalability, compared with the existing algorithms.

In this paper, we propose a new location tracking method for pointer forwarding with distributed HLRs schemes to decrease the location tracking cost. Similar to the distributed HLR schemes, the location information is replicated on all HLRs in our scheme. However, unlike the traditional distributed HLR schemes, we propagate the most recent location changes from an HLR to reset the forwarding pointers in other HLRs, and thus reduce the cost of location tracking for call deliveries. The signaling cost of location propagation may be too costly to be practical. Therefore, we allow some inconsistency to exist among the location data of HLRs and defer the propagation of location changes to reduce the propagation cost. Three delay propagation policies are also proposed and analyzed using simulation. The performance result shows that the delay propagation policies reduce the propagation cost and the propagation resetting process degrades the location tracking cost. Our method preserves a low total cost for call deliveries, registrations and propagations in distributed HLRs.

A low sidelobe two-beam waveguide slot array is designed and measured. The antenna structure should be symmetrical for realizing two symmetrical beams which imposes restriction in slot design for the sidelobe and the gain. The slot coupling distribution is optimized numerically for side-lobe suppression under the condition of the structural symmetry. The first side-lobe level is minimized for the specific antenna efficiency in the continuous source model. This synthesis is reinforced by the full wave slot analysis using the method of moments. The design is confirmed by experiments using a one-dimensional array at 12 GHz and the good agreements between the predictions and the measurements are observed.

This paper describes the evaluation of a fiber-optic reflective displacement sensor that is compensated for variations in light source intensity, pressure, temperature and opacity of ambient medium. Additionally, the distance information is averaged over several points on the target surface, which reduces signal fluctuations due to inhomogeneities. Furthermore, a practical optical fiber reflective sensor model of measuring oil film thickness for thrust bearing is set up in this paper. Actual measurements were made with HEC 3000 tons' thrust bearing and the results were in good agreement with theoretical calculations.

Call admission control (CAC) is becoming vital for multimedia services in the ability of wireless/mobile networks to guarantee Quality of Service (QoS) partially due to the network's limited capacity. In this paper, we propose an optimal call admission control scheme with bandwidth reallocation algorithm (multi-class-CAC-BRA) for multi-classes of adaptive multimedia services in wireless/mobile networks. The multi-class-CAC-BRA approach optimizes revenue for service providers and satisfies QoS requirements for service users. The proposed approach adopts semi-Markov Decision Process to model both call admission control and bandwidth reallocation algorithm. In other words, whenever decisions are made, decisions are made for both call admission control and bandwidth reallocation. Since the non-adaptive multimedia traffic is a special case of the adaptive multimedia traffic, the non-adaptive optimal CAC scheme is a special case of our optimal multi-class-CAC-BRA scheme. Furthermore, the Interior-point Method in linear programming is used to solve the optimal decision problem. Simulation results reveal that the proposed multi-class-CAC-BRA scheme adapts itself well to adaptive multi-class multimedia traffic, achieves optimal revenue, and satisfies QoS requirements that are the upper bounds of handoff dropping probabilities. Our approach solves the optimal adaptive multimedia CAC problem. We believe that this work has both theoretical and practical significance.

Infostations serve a demand for multimedia services with high quality and high data rate. The challenge for Infostation system design is how to download as many information bits when a mobile user drives through an Infostation cell. A variable rate TDMA/TDD radio architecture is chosen for study and experimental implementation because of its flexibility in a multi-user environment. A symbol rate and rate compatible punctured convolutional code controlled adaptive transmission scheme is proposed for the Infostation radio design. Data throughput is analyzed for a mobile user passing through an Infostation cell. The analysis indicates that three data rates at 0.25, 1.0 and 2.0 Mbps achieve greater than 85% of the theoretical throughput while simplifying the implementation. The punctured convolutional code is studied to maximize the Infostation throughput. A recommendation is made to use rate 1/2 coding scheme. An efficient rate adaptation algorithm is proposed and it can track the Rayleigh fading channel well and the capacity loss is negligible in various conditions.

A new speech enhancement technique is proposed assuming that a speech signal is represented in terms of a linear probabilistic process and that a noise signal is represented in terms of a stationary random process. Since the target signal, i.e., speech, cannot be represented by a stationary random process, a Wiener filter does not yield an optimum solution to this problem regarding the minimum mean variance. Instead, a Kalman filter may provide a suitable solution in this case. In the Kalman filter, a signal is represented as a sequence of varying state vectors, and the transition is dominated by transition matrices. Our proposal is to construct the state vectors as well as the transition matrices based on time-frequency pattern of signals calculated by a wavelet transformation (WT). Computer simulations verify that the proposed technique has a high potential to suppress noise signals.

A combination of a software and a systolic hardware implementation for the Quasi Arithmetic compression algorithm is presented. The hardware is implemented as a pipeline hardware implementation. The implementation doesn't change the the algorithm. It just split it into two parts. The combination of parallel software and pipeline hardware can give very fast compression without decline of the compression efficiency.

In this paper, we describe statistical properties of timing jitter of symbol timing recovery circuit for carrierless amplitude/phase modulation (CAP)-based very high-rate digital subscriber line (VDSL) system. Analytical expressions of the timing jitter for envelope-based timing recovery system, such as squarer-based timing recovery (S-TR) and absolute-value-based timing recovery (A-TR) schemes, are derived in the presence of additive white Gaussian noise (AWGN) or far-end crosstalk (FEXT). In particular, the analytical and simulation results of the timing jitter performance are presented and compared for a 51.84 Mb/s 16-CAP VDSL system. The A-TR system implemented digitally meets the DAVIC's VDSL system requirement, which specifies the maximum peak-to-peak jitter value of 1.5 nsec and the acquisition time of 20 msec.

A new method for active suppression of reflected sound waves is proposed in this paper. The proposed control system is based on the state feedback control. FEM (Finite Element Method) was applied to represent the sound field under the system equations as proposed by Samejima et al. A new performance index was derived so as to minimize the sound intensity leaving a control region, which was set around the control source on a wall. On the basis of the system equations and the new performance index, an optimal feedback law governing suppression of waves reflected from the wall was derived. In order to evaluate the validity of the proposed method, computer simulations in one- and two-dimensional sound fields were executed. In a one-dimensional sound field, the time response was examined, and the distribution of the instantaneous sound intensity was evaluated in a two-dimensional sound field. The results showed that the reflected sound waves can be suppressed quite well in one-dimensional sound fields by using this method and that the proposed method can potentially suppress the reflected sound waves in the two-dimensional sound fields as well.