Large parts of the Internet are still incapable of native multicast, and ubiquitous deployment of multicast will take a long time. There are two approaches to provide wide-area multicast service in today's Internet. One is tunneling approach and the other is application-level multicast approach. In this paper, we focus on application-level multicast approach and propose a new scheme which improves the performance penalties of application-level multicast by making use of network support. Because in application-level multicast, endhosts provide multicast functionality instead of routers, application-level multicast delivery tree is inherently less efficient than IP multicast tree. Therefore, in our scheme, the router on the application-level multicast delivery tree alters the tree based on network-level delivery path. We evaluate our scheme with simulation experiment. Our simulation results indicate that our scheme improves the performance of application-level multicast. Further we compare our scheme to the tunneling approach from the viewpoint of transmission performances. The results reveal applicable domains of both approaches.

This letter handles symbolic simulation for high-level hardware design descriptions including uninterpreted functions. Two new heuristics are introduced, which are named "symbolic function table" and "synchronization". In the experiment, the equivalence of a hardware/software codesign was checked up to a given finite number of cycles, which is composed of a behavioral design, that is, a small DSP program written in C, and its register-transfer-level implementation, a VLIW architecture with an assembly program. Our symbolic simulator succeeded in checking the equivalence of the two descriptions which were not tractable without the heuristics.

We propose a novel optical signal processing using an optically pumped vertical-cavity surface-emitting laser (VCSEL) with an external light input. The mode transition between a fundamental and a 1st-high-order transverse mode is induced by an external light injection. Since a single mode fiber (SMF) spatially selects a fundamental transverse mode as an output signal, we are able to realize a nonlinear transfer function, which will be useful in future photonic networks. The mode transition characteristic of a 1.55 µm optically pumped two-mode VCSEL has been simulated by using a two-mode rate equation, which includes the effects of spatial hole burning and spectral hole burning as gain saturation coefficients. We focus on the detuning effect in the injection locking. When the wavelength of an input light with a fundamental mode is slightly longer than that of a VCSEL operating in a 1st-high-order transverse mode, the transverse mode of the VCSEL is switched to a fundamental mode at a critical input power level. This gives us an ideal transfer function for 2R (reamplification and reshaping) regeneration. Also, the proposed scheme may enable polarization insensitive signal processing, which is a unique feature in surface emitting lasers.

This paper presents a scalable and efficient quota-based admission control scheme for the 3rd generation (3G) operator's IP backbone network, where quota denotes a chunk of bandwidth. This research is motivated by the 3G operator's need for guaranteeing end-to-end IP QoS of mobile-to-mobile and mobile-to-server multimedia sessions. In the proposed scheme, the quota size of a path implies the proper amount of allocated and released resources on the path condition. Employing the quota size makes the job of allocating or releasing resources at nodes in a path simple so that it becomes scalable. Moreover, with this simple scheme, an edge node can be allowed not only to initiate the allocation/release request but also to perform admission control function. To maximize the efficiency, the path quota size varies depending on the bottleneck link condition in the path. In high offered load, the proposed scheme decreases the path quota size and retains higher utilization while it requires lower signaling cost than the fixed scheme using a fixed size aggregation. As the load lessens, it increases the path quota size and reduces the signaling cost significantly.

Software applications for the transfer of money or personal information are increasingly common on the Internet. These applications require user authentication for confirming legitimate users. One-time password authentication methods risk a stolen-verifier problem or other steal attacks because the authentication on the Internet server stores the user's verifiers and secret keys. The SAS-2 (Simple And Secure password authentication protocol, ver.2) and the ROSI (RObust and SImple password authentication protocol) are secure password authentication protocols. However, we have found attacks on SAS-2 and ROSI. Here, we propose a new method which eliminates such problems without increasing the processing load and can perform high security level same as S/Key systems without resetting the verifier.

This paper proposes a new correction technique for a linear amplification with nonlinear components (LINC) transmitter. The technique, which is based on the minimum mean squared error (MMSE) criterion, estimates the gain and phase imbalance between the two amplifier branches. With information on the estimation, the imbalance is offset by controlling the amplitude and phase of the input signal that is fed into one of the two amplifiers. Computer simulations with a DS-CDMA system demonstrate that this method can compensate for the imbalance and sufficiently suppress the out-of-band distortion spectrum.

There is an increasing demand for the technology of content distribution, by which each user can request desired content through a network. Because of the low efficiency of existing systems, we proposed a new block transfer type video distribution system called Burst VoD. The Burst VoD system aggressively utilizes multicasting, and divides the content data into a mass of block files, which it periodically transmits to a terminal through a high-speed network, using a higher rate than the playback speed. However, by using the scheduling algorithm of the Burst VoD system, when users request the same content from different periods, the VoD server repeatedly transmits the same block files in different periods. In this paper, we propose an advanced scheduling algorithm based on the Burst VoD system to improve its multicasting efficiency. In addition, we propose a multi-channel BurstVoD in order to reduce the interface bandwidth of client.

A new mobility management architecture is proposed to optimize end-to-end routes for mobile nodes (MNs) and mobile routers (MRs) within a nested mobile network environment. By applying local network mobility management mechanisms based on Hierarchical Mobile IPv6 (HMIPv6) to a mobile network, the proposed approach can optimize the route to the mobile network effectively. Combining the proposed route optimization methods and HMIPv6 functionality can enable it to provide more effective route optimization, reducing the burden of location registration for handovers. A route optimization method for local fixed nodes in a mobile network has also been developed by adding proxy mobile node and correspondent node functions to the MRs. Numerical evaluations on mean route length and traffic routed through network nodes demonstrate the effectiveness and applicability of the proposed methods especially in large-scale networks.

The handoff in Mobile IP networks causes packet sequence disruption during a packet forwarding procedure and may result in performance degradation in higher layer protocols. We investigate the impact of handoff in the Mobile IPv6 networks, where an optimized routing with the smooth handoff is adopted. The impact on the packet sequence is measured by an 'unstable time period (UTP)' and a 'silence time period (STP).' The UTP explains the time duration of out-of-sequence packets while the STP reflects the blackout duration of a mobile node after the initiation of handoff procedure. With the analysis on the UTP and STP, the total transient time period (denoted as handoff time period or HTP) after the handoff initiation can be estimated. In our previous work, focusing on the UTP, the packet flow sequence under the smooth handoff is analyzed for the Mobile IPv4 networks. The proposed queuing-based analysis is extended in this work for the Mobile IPv6 networks. That is, several modifications are made to conform to Mobile IPv6 and at the same time the queuing analysis itself is improved to better model the handoff procedure. The numerical results show that the queuing delay for the handoff packets (affected by background traffic) and the involved link (or route) capacities affect the estimated UTP, STP, and HTP. In addition, two schemes such as priority queuing and buffered packet forwarding are introduced to reduce the transient period and the improvements are analyzed for comparison.

This letter considers multicarrier CDMA systems using rectangular chip waveform, where each user's data stream is serial to parallel converted to a number of lower rate streams. Each lower rate stream is then spread by a random spreading code and a suitable chip rate before modulating orthogonal carriers. It is shown that, for a fixed system bandwidth and a given number of carriers, there exists an optimal carrier spacing that minimizes the multiple access interference. Numerical examples also show that the multicarrier DS-CDMA system previously proposed in performs very close to the muticarrier CDMA system using the optimal carrier spacing.

This paper investigates modified random timers based on uniform and exponentially distributed timers for feedback scalability for large groups. We observe the widely-used probability distribution functions and propose new ones that are aware of network delays. The awareness of network delays of our proposed modified p.d.fs proves to be able to achieve lower expected number of messages compared to the original ones given that the parameters are optimized for the network variables: the number of receivers, and the network delay. In our analysis we derive an equation to estimate the optimized parameter based on these network variables. We also simulate the p.d.fs for heterogenous network delays and find that each receiver only needs to be aware of its network delay.

In this letter, we propose an enhanced time-based registration method and analyze the performance numerically. In the analysis, we assume Poisson call arrival distribution and exponential cell resident time. The performance of the enhanced time-based registration method is compared with the performance of the original time-based registration method. In the comparisons, we see that in a certain range of parameters, the enhanced time-based registration method has better performance.

We propose a new protocol to achieve fault recovery of multicast applications in IP internetwork with mobile participators. Our protocol uses the basic unicast routing capability of IETF Mobile IP as the foundation, and leverages existing IP multicast models to provide reliable multicast services for mobile hosts as well. We believe that the resulting scheme is simple, scalable, transparent, and independent of the underlying multicast routing facility. A key feature of our protocol is the use of multicast forwarding agent (MFA) to address the scalability and reliability issues in the reliable mobile multicast applications. Our simulation results show the distinct performance advantages of our protocol using MFAs over two other approaches proposed for the mobile multicast service, namely Mobile Multicast Protocol (MoM) and bi-directional tunneling, particularly as the number of mobile group members and home agents (HAs) increases.

The multimedia applications have recently generated much interest in wireless network infrastructure with supporting the quality-of-service (QoS) communications. In this paper, we propose a lantern-tree-based QoS on-demand multicast protocol for wireless ad hoc networks. Our proposed scheme offers a bandwidth routing protocol for QoS support in a multihop mobile network, where the MAC sub-layer adopts the CDMA-over-TDMA channel model. The QoS on-demand multicast protocol determines the end-to-end bandwidth calculation and bandwidth allocation from a source to a group of destinations. In this paper, we identify a lantern-tree for developing the QoS multicast protocol to satisfy certain bandwidth requirement, while the lantern-tree is served as the multicast-tree. Our lantern-tree-based scheme offers a higher success rate to construct the QoS multicast tree due to using the lantern-tree. The lantern-tree is a tree whose sub-path is constituted by the lantern-path, where the lantern-path is a kind of multi-path structure. This obviously improves the success rate by means of multi-path routing. In particular, our proposed scheme can be easily applied to most existing on-demand multicast protocols. Performance analysis results demonstrate the achievements of our proposed protocol.

In this paper, we discuss the performance of a basic scheme to support network mobility. Network mobility arises when an entire network segment, such as a network inside a vehicle, changes its topological location and thus its access point to the fixed backbone network. Mechanisms to support network mobility are necessary to maintain sessions. The approach followed by the IETF (NEMO Basic Support) and us (B-ORC) is to establish a bi-directional tunnel between the mobile network and the Internet. As we show, this bi-directional tunnel is a performance bottleneck and leads to single points of failure. In order to address the issues of the existing mobile network architecture, we propose enhanced operations of the basic mobile network protocol to achieve reliability and efficiency: (1) multiple bi-directional tunnels between the mobile network and the Internet, and (2) policy-based routing. The proposed operations could be realized by extending the existing architecture and protocol. The performance of various multihoming configurations is evaluated based on the implementation of our own basic scheme. The evaluation criteria are delay, throughput and latency. The results are encouraging and show we can achieve a better throughput.

The main issue in real time voice applications over Internet is to investigate a lossless playout without jitter while maintaining playout delay as small as possible. Existing playout algorithms estimate network delay by using timestamps and determine the playout schedule only at the beginning of each talkspurt. Also their scheduled playout time is determined based on a fixed upper playout delay bound over a talkspurt. The sliding adaptive playout algorithm we propose can estimate jitter without using timestamps and its playout time is allowed to slide to a certain extent. The aim of sliding playout schedule is to determine the scheduled playout time at the beginning of each talkspurt based on the playout delay expected under the normal condition where the degree of actual jitter is below the magnitude which is not quite large in relation to variations in the "baseline" delays. Then the proposed algorithm can be effectively applied to minimize the scheduled playout delay while improving the voice quality against a spike which may occur at the beginning of a talkspurt as well as a spike which occurs in the middle of a talkspurt. We develop an analytical model of the general adaptive playout algorithms and analyze the playout buffer performance for different degrees of jitter, different values of the scheduled playout delay, different maximum lengths of delay spikes, and arbitrary tolerable ranges of sliding. Based on the analytical results, we suggest the specific values of parameters used in the sliding algorithm.

The 6to4 method enables separate IPv6 sites to connect to the IPv6 Internet via a 6to4 relay router without an explicit IPv6-over-IPv4 tunnel setup. There are about a dozen open 6to4 relay routers worldwide but none of these have been installed in Japan. We therefore decided to evaluate the 6to4 mechanism and set ourselves the goal of improving the 6to4 operation within Japan. To accomplish this, in March 2002, we installed an open 6to4 relay router in Japan with the cooperation of the WIDE project and started this experiment. This paper describes our experiment and analysis of IPv6 traffic through our 6to4 relay router, as well as considerations derived from our experiment.

Two fast quarter-pixel motion estimation algorithms using the integer motion vector (MV) already obtained by any fast motion estimation algorithm are proposed, which perform MV search only in half-pixel and quarter-pixel unit. The proposed algorithms search MV about 4-9 times as fast as the full search in quarter-pixel unit, while the PSNR is slightly degraded.

FEC (Forward Error Correction) can repair the damage to communication quality due to packet loss. The growing requirement of FEC for high-quality video transmission is inevitable on broadband networks. We have designed and implemented FEC, and integrated it to our developed video transmission system named "mpeg2ts." Our goal is to make it possible to deploy this system on the broadband Internet. However, the problem with constant redundancy of FEC is that weakness to fluctuation of network condition. To resolve this problem, in this paper, we propose and evaluate an efficient FEC method for high-quality video transmission. The proposed mechanisms can provide robustness as well as saving of processing load and optimization of bandwidth consumption. Moreover, we integrate it into a system to deploy it on the real broadband Internet. Transmission experiment demonstrates availability of developed system deployed on the network.

This paper deals with the statistical modeling of a Time-Frequency Series of Speech (TFSS), obtained by Short-Time Fourier Transform (STFT) analysis of the speech signal picked up by a linear microphone array with two elements. We have attempted to find closer match between the distribution of the TFSS and theoretical distributions like Laplacian Distribution (LD), Gaussian Distribution (GD) and Generalized Gaussian Distribution (GGD) with parameters estimated from the TFSS data. It has been found that GGD provides the best models for real part, imaginary part and polar magnitudes of the time-series of the spectral components. The distribution of the polar magnitude is closer to LD than that of the real and imaginary parts. The distributions of the real and imaginary parts of TFSS correspond to strongly LD. The phase of the TFSS has been found uniformly distributed. The use of GGD based model as PDF in the fixed-point Frequency Domain Independent Component Analysis (FDICA) provides better separation performance and improves convergence speed significantly.