Radio waves propagating through tunnels are strongly attenuated in the presence of discontinuities such as bends and branches. The useful structural modifications are requested to get better circumstances for radio waves in tunnels. In this paper, we propose several modifications arranged in a conventional T-junction of two-dimensional tunnels and analyze the transmission characteristics of radio waves by using the finite volume time domain (FVTD) method.

Second harmonic signal feedback technique is applied to an HBT power amplifier for Wide-band CDMA (W-CDMA) mobile communication system to improve its linearity and efficiency. This paper describes the feedback effect of the 2nd harmonic signal from the output of the amplifier to the input on the 3rd order intermodulation distortion (IMD) products and Adjacent Channel leakage Power (ACP) of the power amplifier. The feedback amplifier, using an InGaP/GaAs HBT with 48 fingers of 3 20 µ m emitter, exhibits a 10 dB reduction in the level of the 3rd order IMD products. In addition, an ACP improvement of 7 dB for the QPSK modulation signal with a chip rate of 4.096 Mcps at 1.95 GHz was realized. As a result, the amplifier achieves a power-added efficiency of 41.5%, gain of 15.3 dB, and ACP of 43.0 dBc at a 5 MHz offset frequency and output power of 27.5 dBm. At the output power of 28 dBm, the power-added efficiency increases to 43.3% with an ACP of 40.8 dBc.

We have demonstrated the single-chip RF front-end GaAs MMIC for the Japanese Personal Handy-phone System. It has a high efficiency HPA, a T/R switch, a LNA and a low-distortion down converter mixer. The IC employs a negative voltage generator for use of single voltage DC power supply. The HPA provides an output power of 21.5 dBm, with an ACPR of 55 dBc and an efficiency of 35%. The LNA has a noise figure of 1.6 dB and a gain of 14 dB with current of 2.3 mA. The newly developed active cascode FET mixer has a high IIP3 of 1 dBm with a high conversion gain of 10 dB and low consumption current of 2.3 mA. The IC is characterized by high performance for RF front-end of PHS handheld terminals. The IC is available in a 7.0 mm6.4 mm1.1 mm plastic package.

This letter proposes improvement to the previously presented watermarking method which spreads an ID pattern with a random sequence and embeds it throughout the spatial domain of an image. The proposed method can extract embedded watermarks without an original image even from images converted by brightness/contrast conversion, edge-enhancement, posterization and JPEG compression.

Following our former works on regular sampling in wavelet subspaces, the paper provides two algorithms to estimate the truncation error and aliasing error respectively when the theorem is applied to calculate concrete signals. Furthermore the shift sampling case is also discussed. Finally some important examples are calculated to show the algorithm.

In this paper, a new multicast routing method for layered streams is proposed. This method is an extension of the weighted greedy algorithm (WGA) and uses two kinds of weight values to refine the link distance. It can cope with dynamic change in the group members without multicast tree re-construction. The method is compatible with the RSVP and can be utilized in existing shared tree type routing protocols such as CBT and PIM sparse mode. The network resources can be utilized efficiently; furthermore, the loss rate of member's requests to receive more layers can be reduced by this routing method when a sufficient number of nodes have the packet filtering function and a sufficient number of hops is permitted.

This paper presents a novel distortion compensation technique using an active inductor. First, we describe the input-reflection-coefficient characteristics of a GaAs MESFET active inductor when input power increases. We show that the inductor exhibits positive amplitude deviation and negative/positive phase deviation as the input power increases when the biases of the FETs are set appropriately. The chip size of the fabricated active inductor is less than 0.52 mm2. Then, we show that third-order intermodulation is improved when the active inductor is used as a predistortion linearizer. Third-order intermodulation was improved over the output range from 14 dBm to 25 dBm, and at the output of 15 dBm, third-order intermodulation was improved approximately by 9 dB when the predistortion linearizer was introduced. The active inductor can thus function as a miniaturized predistortion linearizer by using it in the input matching circuit of a power amplifier. This technique can be applied in the miniaturization of wireless communication devices.

A novel method to enhance the practical security of interferometric quantum cryptography is proposed, giving the protocol and detailed constructions including a controlled spontaneous photon emitter, a superradiance amplifier, beam splitters, phase shifters, and a pair of Mach-Zehnder interferometers. The intrinsic uncertainty due to the random phase selection out of three, leads to the detection of eavesdropping. The physical uncertainty of the controlled spontaneous emission of coherent photons also adds temporal equivocation to confuse eavesdroppers.

We first demonstrate a self-pulsation phenomenon in a semiconductor ring laser(SRL). Not only self-mode-locked optical pulse but self-Q-switched optical pulse can be observed in a SRL. Furthermore, experimental results show that the repetition period of the Q-switched optical pulse train can be controlled by the injection current to a SRL.

This paper evaluates the performance of a reliable multicast protocol for bulk-data transfer over unreliable networks via IP-multicast. Bulk-data type reliable multicast appears promising for commercial publishing and large-scale data replication. The proposed reliable multicast transport protocol (RMTP) provides high-performance due to the use of IP multicast while also providing confirmed and error free transfer by end-to-end controls. The protocol includes a multi-round selective repeat scheme dedicated for bulk-data multicast applications. This paper examines the multicast retransmission procedures in RMTP through analysis and tests on an implemented system and clarifies the basic performance behavior of the protocol. Evaluations are conducted with regard to retransmission redundancy, transfer time, and packet processing load with various error conditions and number of receivers. Against the response concentration problem seen in end-to-end communication, the backoff time algorithm is applied to the protocol; the limits it places on system scalability are clarified.

We proposed Reversible Energy Recovery Logic (RERL) using an 8-phase clocking scheme, which is a dual-rail reversible adiabatic logic for ultra-low-energy applications. Because we eliminated non-adiabatic energy loss in RERL by using the concept of reversible logic, RERL has only adiabatic and leakage losses. In this paper we explain its operation and logic design and present its simulation and experimental results. We also present an energy-efficient 8-phase, clocked power generator that uses an off-chip inductor. With simulation results for the full adder, we confirmed that the RERL circuit consumed substantially less energy than other logic circuits at low-speed operation. We evaluated a test chip implemented with a 0.6-µm CMOS technology, which integrated a chain of inverters with a clocked power generator. In the experimental results, the RERL circuit consumed only 4.5% of the dissipated energy of a static CMOS circuit at an optimal operating speed of 40 kHz. In conclusion, RERL is suitable for the applications that do not require high performance but low-energy consumption because its energy consumption can be decreased to the minimum by reducing the operating frequency until adiabatic and leakage losses are equal.

A timing synchronization is required in the mobile station to determine the correct transmission timing of the mobile-to-base bursts. In this letter, a timing synchronization technique using the reliability check and smoothing algorithm is proposed for the GSM receiver. The reliability check scheme extends the usefulness of this algorithm into low SNR region. And also smoothing algorithm is carried out by a first-order filter with an asymmetric step size. Simulation results show that the proposed algorithm is adequate for timing recovery of GSM modem.

We show by numerical calculations that a chaotic neuron model driven by a weak sinusoid has resonance. This resonance phenomenon has a peak at a drive frequency similar to that of noise-induced stochastic resonance (SR). This neuron model was proposed from biological studies and shows a chaotic response when a parameter is varied. SR is a noise induced effect in driven nonlinear dynamical systems. The basic SR mechanism can be understood through synchronization and resonance in a bistable system driven by a subthreshold sinusoid plus noise. Therefore, background noise can boost a weak signal using SR. This effect is found in biological sensory neurons and obviously has some useful sensory function. The signal-to-noise ratio (SNR) of the driven chaotic neuron model is improved depending on the drive frequency; especially at low frequencies, the SNR is remarkably promoted. The resonance mechanism in the model is different from the noise-induced SR mechanism. This paper considers the mechanism and proposes possible explanations. Also, the meaning of chaos in biological systems based on the resonance phenomenon is considered.

We consider a system of coupled two oscillators with external force. At first we introduce the symmetrical property of the system. When the external force is not applied, the two oscillators are synchronized at the opposite phase. We obtain a bifurcation diagram of periodic solutions in the coupled system when the single oscillator has a stable anti-phase solution. We find that the synchronized oscillations eventually become in-phase when the amplitude of the external force is increased.

Auctions are a critical element of the electronic commerce infrastructure. But for real-time applications, auctions are a potential problem - they can cause significant time delays. Thus, for most real-time applications, sealed-bid auctions are recommended. But how do we handle tie-breaking in sealed-bid auctions? This paper analyzes the use of multi-round auctions where the winners from an auction round participate in a subsequent tie-breaking second auction round. We perform this analysis over the classical first-price sealed-bid auction that has been modified to provide full anonymity. We analyze the expected number of rounds and optimal values to minimize communication costs.

The main issues of the next generation Internet are high performance multicast, security and QoS (Quality of Service). The network layer multicast is accepted as a powerful solution to reduce communication overhead in networks as well as hosts. Although a lot of multicast routing protocols have been developed and have worked for the current Internet, they have drawbacks in scalability, reliability and network load balancing. In this paper, we propose an enhanced multicast routing protocol called DCBT (Decentralized Core based Tree), which includes some features of CBT (Core Based Tree), PIM-SM (Protocol Independent Multicast-Sparse Mode) and BGMP (Border Gateway Multicast Protocol). Fundamentally DCBT differs from those in that it covers not only intra-region but also inter-region multicast. RNs (Regional networks) have their intra-region shared trees, which are interconnected by an inter-region shared tree for a multicast group. New two concepts are introduced: the one is RBTS (Region Based Tree Switching) to achieve network load balancing and the other is to redefine TOS (Type Of Service) byte to make special multicast routing decision. In addition, the effect of core location is given by changing its position within an RN. Finally we provide the performance evaluation of DCBT by OPNET network simulator over a real scale network model.

IP telephony systems are expected to be deployed worldwide in the near future because of their potential for integrating the multimedia communication infrastructure over IP networks. Phone-to-phone connection over an IP network via IP telephony gateways (IP-GWs) is a key feature of the system. In an IP telephony system, a low-bit-rate voice codec is used to improve bandwidth efficiency. However, due to the packet transfer method over the IP network, it is necessary to add packet headers, including IP, UDP, and RTP headers, which increases the header overhead and thus decreases transfer efficiency. Moreover, because there will be large numbers of short voice packets flowing into the IP network, the load on the Internet will increase. We propose voice stream multiplexing between IP-GWs to solve these problems. In this scheme, multiple voice streams are connected between a pair of IP-GWs, enabling multiplexed voice stream transfer. The voice stream multiplexing mechanism can reduce the header overhead as well as decrease the number of voice packets. The voice stream multiplexing we propose is to concatenate RTP packets destined for the same IP-GW at a multiplexing interval period into a single UDP packet. The advantage of this method is that no new additional header is required and the current well-defined H. 323 and RTP standards can be applied with minimum changes. We implemented and tested the system. The results show that the proposed method is effective at reducing both the header overhead and the number of packets. In a typical case, the bandwidth is cut by 40% for eight G. 723.1-encoded voice streams through header overhead reduction and the number of voice packets is also decreased to 1/8. Furthermore, this method can easily be enhanced to a general RTP packet multiplexing method that is applicable not only to an IP-GW but also to other RTP multiplexing and de-multiplexing applications.

Network applications such as FTP, WWW, Mirroring etc. are presently operated with little or no knowledge about the characteristics of the underlying network. These applications could operate more efficiently if the characteristics of the network are known and/or are made available to the concerned application. But network characteristics are hard to come by. The IP Performance Metrics working group (IETF-IPPM-WG) is working on developing a set of metrics that will characterize Internet data delivery services (networks). Some tools are being developed for measurements of these metrics. These generally involve active measurements or require modificationsin applications. Both techniques have their drawbacks. In this work, we show a new and more practical approach of estimating network characteristics. This involves gathering and analyzing the network's experience. The experience is in the form of traffic statistics, information distilled from management related activities and ubiquitously available logs (squid access logs, mail logs, ftp logs etc. ) of network applications. An analysis of this experience provides an estimate of the characteristics of the underlying network. To evaluate the concept we have developed and experimented with a system wherein the network characteristics are generated by analyzing the logs and traffic statistics. The network characteristics are made available to network clients and administrators by Network Performance Metric (NPM) servers. These servers are accessed using standard network management protocols. Results of the evaluation are presented and a framework for efficient operation of network operations, using the network characteristics is outlined.

Certain relationships between successive packet loss patterns and packet spacing are described. To observe a successive packet loss pattern, one possible method is to investigate test packets that are generated at certain intervals (e. g. , Poisson interval, constant regular interval). However, successive packet loss strongly depends on the interval generated. If test packets are generated with long intervals, then successive loss pattern cannot be shown. Thus, in such a method, where the packet intervals may sometimes be long or short, the successiveness of the packet loss should be considered in terms of the packet spacing. To clarify the relationship between the successive packet loss and the packet spacing, we analyze data based on observation of an actual network with the loss window size as a parameter. We find that when the packet spacing is narrower, i. e. , has a shorter interval, the probability becomes higher that the packet immediately following a single packet loss would also be lost.

As network services become more diverse and powerful, service applications that perform such services are acquiring an ever-larger amount of complicated and changeable relationships. We present a network dependence graph (NDG) that captures both data and control flow relationships between components of service applications that work collaboratively. This graph is constructed based on analysis of both the behavior of each of the service applications and their configuration, which describes the device names they refer to, and allows network slicing to be implemented as a simple graph traversal. Network slicing is the extraction of necessary and minimum service components that may affect the execution of a specified service application; it helps a network manager to find the location of service faults lurking somewhere in the network. We also present a method for locating faults that uses network slicing and a system based on this method.