With the proliferation of various types of computing and networking resources in ubiquitous computing environments, an architecture allowing mobile users to flexibly access these resources is desirable. We have focused our attention on the access link resources of devices surrounding users. Our framework named Personal Mesh allows personal devices to seamlessly access the Internet via appropriate access links available in a personal area network. The Personal Mesh deals with two technical issues: access link selection management and a PAN mobility support mechanism. In this paper, we describe the design and implementation of Personal Mesh and show the effectiveness of our system by experiment.

Layered multicast approach enables IP multicast to adapt to heterogeneous networks. In layered multicast, each layer of a session is sent to separate multicast groups. These layers will be transmitted on the same route, or on different routes. However, traditional congestion control schemes of layered multicast do not consider the case when layers of a session are transmitted on different routes. In this paper, at first we show that in sparse-mode routing protocols like PIM-SM and CBT, layers of a session can be mapped to different Rendezvous Points or cores due to the bootstrap mechanism. It means that layers of a session can be transmitted on different routes. We then show that traditional congestion control schemes of layered multicast do not work properly in sparse-mode routing regions. At last we introduce Rendezvous Point based Layered Multicast (RPLM), a novel congestion control scheme suitable for sparse-mode routing regions, and show that RPLM works efficiently in regions using sparse mode routing protocols. RPLM uses per-RP packet loss rate instead of the overall one to detect congestion on each route, and can react to congestion quickly by dropping the highest layer on the congested route. In addition, RPLM simultaneously drops all the layers those are useless in quality's improvement to prevent bandwidth waste.

This paper presents a data gathering protocol for wireless sensor network applications that require high throughput and topology adaptability under the premises of uniform traffic and energy-rich environments. Insofar as high throughput is concerned, TDMA is more suitable than CSMA. However, traditional TDMA protocols require complex scheduling of transmission time slots. The scheduling burden is the primary barrier to topology adaptability. Under the premises of uniform traffic and energy-rich environments, this paper proposes a token-scheduled multi-channel TDMA protocol named TKN-TWN to ease the scheduling burden while exploiting the advantages of TDMA. TKN-TWN uses multiple tokens to arbitrate data transmission. Due to the simplified scheduling based on tokens, TKN-TWN is able to provide adaptability for topology changes. The contention-free TDMA and multi-channel communication afford TKN-TWN the leverage to sustain high throughput based on pipelined packet forwarding. TKN-TWN further associates the ownership of tokens with transmission slot assignment toward throughput optimization. We implement TKN-TWN on Tmote Sky with TinyOS 2.1.1 operating system. Experimental results in a deployed network consisting of 32 sensor nodes show that TKN-TWN is robust to network changes caused by occasional node failures. Evaluation also shows that TKN-TWN is able to provide throughput of 9.7KByte/s.

In this paper, we introduce a concept of minimum bandwidth guaranteed service model for mobile multimedia. In this service model, service is defined in the context of the guaranteed minimum bandwidth and the residual service share. Each flow under this service model is guaranteed with its minimum bandwidth and provided with more in proportion to the residual service share if there is leftover bandwidth. The guaranteed minimum bandwidth assures a flow to keep minimum tolerable quality regardless of the network load, while the leftover bandwidth enhances the quality of service according to the application's adaptivity and the user's interest. We show that the minimum bandwidth guaranteed service model could be implemented by a two-folded wireless packet scheduler consisting of a guaranteed scheduler and a sharing scheduler. Wireless channel condition of each flow is considered in scheduling so that wireless resource can be distributed only to the flows of good channel state, improving total wireless link utilization. We evaluate the service model and the scheduling method by simulation and implementation.

The Communications Research Laboratory (CRL) started a new project named the New Generation Mobile Network Project in April 2002. The target of this project is the development of new technologies to enable seamless and secure integration of various wireless access networks such as 3rd and 4th generation cellular, wireless LAN, high-speed mobile wireless, wired communications, and broadcasting networks. This paper presents an overview of CRL's new generation mobile communication system that is called The Multimedia Integrated Network by Radio Access Innovation Plus (MIRAI+), as well as details the role of Software Radio Technology (SDR) in MIRAI+.

This paper presents a distributed power control scheme for next-generation multiservices CDMA systems. CDMA has inherent capability to control the quality-of-service (QoS) requirements by assigning different power levels to each traffic type. Toward this, optimum power control schemes have been investigated. The main drawback of the previously proposed algorithms is that they would require all users' transmission state necessiating a complicated control process or peak-rate badnwidth allocation. To overcome this, we exploit the Markovian property to obtain the statistics of the traffic. The statistical formulation is presented for allocating power distributedly so as to keep the "collision" probability below a predefined probability. Numerical examples show that the distributed power control scheme allows better utilization of wireless resources through statistical multiplexing than peak-rate bandwidth assignment, and it does not require a complicated control process while keeping total transmitted power at slightly greater than optimum power control.

We describe an approach for modelling a person's face for model-based coding. The goal is to estimate the 3D shape by combining the contour analysis and shading analysis of the human face image in order to increase the quality of the estimated 3D shape. The motivation for combining contour and shading cues comes from the observation that the shading cue leads to severe errors near the occluding boundary, while the occluding contour cue provides incomplete surface information in regions away from contours. Towards this, we use the deformable model as the common level of integration such that a higher-quality measurement will dominate the depth estimate. The feasibility of our approach is demonstrated using a real facial image.

Fairness as defined in wireline network cannot be achieved in wireless packet networks due to the bursty and location-dependent channel errors of wireless link. Channel-state dependent scheduling and compensation mechanism for errored flows are generally employed to improve the fairness in wireless packet scheduling algorithms. Most of the wireless scheduling algorithms, however, have two common significant problems. One problem is that they operate incorrectly unless all flows have the same packet size. This is due to the incorrect lead-and-lag model and the swapping-based rough compensation mechanism of the algorithms. The other problem is the degradation of error-free flow during compensation. The root of the degradation is that the bandwidth for compensation cannot be reserved since it is very difficult to predict. In this paper, we introduce WGPS (Wireless General Processor Sharing) to define fairness in wireless networks and present PWGPS as a packetized algorithm of WGPS. In our method, lead and lag of all flows are defined in terms of virtual time which is flexible enough to deal with packet size differences. Further, compensation is done by increasing the service share of the errored flow to a predetermined amount Δ. This enables the maximum bandwidth for compensation can be estimated and thus reserved. Our method can be proved to achieve improved fairness as compared with the previous scheduling algorithms.

In the future network, optical technology will play a stronger role not only for transmission but also for switching. Optical burst switching (OBS) emerged as a promising switching paradigm. It brings together the complementary strengths of optics and electronics. This paper presents the design and implementation of an overlay mode burst-switched photonic network testbed, including its architecture, protocols, algorithms and experiments. We propose a flexible "transceiver + forwarding" OBS node architecture to perform both electronic burst assembly/disassembly and optical burst forwarding. It has been designed to provide class of service (CoS), wavelength selection for local bursts, and transparency to cut-through bursts. The functional modules of OBS control plane and its key design issues are presented, including signaling, routing, and a novel scheduling mechanism with combined contention resolution in space and wavelength domains. Finally, we report the experimental results on functional verification, performance analysis and service demonstration.

The Reed-Solomon code is a versatile channel code pervasively used for communication and storage systems. The bit-serial Reed-Solomon encoder has a simple structure, although it is somewhat difficult to understand the algorithm without considerable theoretical background. Some professionals and students, not able to understand the algorithm thoroughly, might need to implement the bit-serial encoder for themselves. In this letter, a step-by-step method is presented for the implementation of the bit-serial encoder even without understanding the internal algorithm, which would be helpful for VHDL, DSP, and simulation programming.

We investigate and propose the utilization of regenerative and non-regenerative relaying terminals in downlink cooperative MIMO communications, such as in base-station/router-relay-user transmission under different schemes. The source is equipped with multiple antennas, while the relays and destination are single-antenna terminals. From the source to the relays, symbols are transmitted using MIMO spatial-multiplexing technique. Depending on the type of relaying scheme, the relays either fully decode or amplify the received signal before retransmitting it to the destination using simple TDM transmission or Alamouti's space-time coding. We show that the proposed system realizes MIMO performance in single-antenna system environment, and performance-wise it is superior to existing transmission schemes, especially in low-SNR conditions. Furthermore, the proposed system is shown to give a diversity order of N-M+1, similar to that of MIMO V-BLAST system.

This paper describes the details of the iteration process used to determine the transfer functions of linear time-invariant (LTI) circuits causing the memory effect of power amplifier (PA). An outline of the method is reported in our work presented at ICCS2012. The accuracy of the method is improved by using cross-correlation spectra at three signal levels, and its validity is confirmed by a computer simulation. The method can be applied to online updating of PAs operating in mobile communication systems. The updating is realized separately from the fast varying nonlinear coefficients. The possibility of updating with a short interval is indirectly shown for the nonlinear coefficients using a procedure similar to that of memoryless PAs. For PAs characterized by the method, this paper also describes the inverses that cancel the nonlinear distortion with minimum complexity. The validity of the inverse is confirmed by a computer simulation on the power spectrum of the PA for orthogonal frequency-division multiplexing (OFDM) signals with 500 subcarriers. The simulated spectra show that the fifth order or higher inverses are effective in keeping adjacent channel leakage power ratio (ACLR) lower than -60dB at the practical signal level. Improvements in the error vector magnitude (EVM) due to the inverse were also confirmed by reductions of gain and phase variations under varying envelope conditions.

System capacity of a wireless system can be improved greatly by using variable rate transmission. Assuming a low-rate and wide-coverage signaling-only wireless network, in this paper we evaluate, analytically and numerically, the extent of this improvement for various schemes with variable transmission rates. We considered log-normal shadowing as well as the effect of Rayleigh fading. Simulation results show close proximity with the analytical predictions.

Concurrent transmission (CT) is a revolutionary multi-hop protocol that significantly improves the MAC- and network-layer efficiency by allowing synchronized packet collisions. Although its superiority has been empirically verified, there is still a lack of studies on how the receiver survives such packet collisions, particularly in the presence of the carrier frequency offsets (CFO) between the transmitters. This work rectifies this omission by providing a comprehensive evaluation of the physical-layer receiver performance under CT, and a theoretical analysis on the fading duration of the beating effect resulting from the CFO. The main findings from our evaluations are the following points. (1) Beating significantly affects the receiver performance, and an error correcting mechanism is needed to combat the beating. (2) In IEEE 802.15.4 systems, the direct sequence spread spectrum (DSSS) plays such a role in combatting the beating. (3) However, due to the limited length of DSSS, the receiver still suffers from the beating if the fading duration is too long. (4) On the other hand, the basic M-ary FSK mode of IEEE 802.15.4g is vulnerable to CT due to the lack of error correcting mechanism. In view of the importance of the fading duration, we further theoretically derive the closed form of the average fading duration (AFD) of the beating under CT in terms of the transmitter number and the standard deviation of the CFO. Moreover, we prove that the receiver performance can be improved by having higher CFO deviations between the transmitters due to the shorter AFD. Finally, we estimate the AFD in the real system by actually measuring the CFO of a large number of sensor nodes.

We study the properties of information dissemination over location-aware gossiping networks leveraging location-based real-time communication applications. Gossiping is a promising method for quickly disseminating messages in a large-scale system, but in its application to information dissemination for location-aware applications, it is important to consider the network topology and patterns of spatial dissemination over the network in order to achieve effective delivery of messages to potentially interested users. To this end, we propose a continuous-space network model extended from Kleinberg's small-world model applicable to actual location-based applications. Analytical and simulation-based study shows that the proposed network achieves high dissemination efficiency resulting from geographically neutral dissemination patterns as well as selective dissemination to proximate users. We have designed a highly scalable location management method capable of promptly updating the network topology in response to node movement and have implemented a distributed simulator to perform dynamic target pursuit experiments as one example of applications that are the most sensitive to message forwarding delay. The experimental results show that the proposed network surpasses other types of networks in pursuit efficiency and achieves the desirable dissemination patterns.

Decentralized and unstructured peer-to-peer (P2P) networks such as Gnutella are attractive for large-scale information retrieval and search systems because of their scalability, fault-tolerance, and self-organizing nature. Because of this decentralized architecture, however, traditional P2P keyword search systems are difficult to globally share useful semantic knowledge among nodes. As a result, traditional P2P keyword search systems cannot support semantic search (support only naive text-match search). In this paper, we describe a design of the semantic P2P keyword search system. We exploit the semantics of correlation among keywords rather than synonym. The key mechanism is query expansion, where a received query is expanded based on keyword relationships. Keyword relationships are improved through search and retrieval processes and each relationship is shared among nodes holding similar data items. This semantic P2P search system has two main advantages. First, expanding search results through query expansion increases the possibility of locating desired data items which would not be found by traditional P2P search systems due to the keywords' textual mismatch. Second, keyword relationships originally introduced for query expansion, can be used for result ranking. Our main challenges are 1) managing keyword relationships in a fully decentralized manner and 2) maintaining the quality of search results, while suppressing result implosion. We also describe the prototype implementation and evaluation of the semantic P2P search system.

We have designed a scalable and expressive naming system called SENS, capable of retrieving information of computing and content resources distributed widely across the Internet through exact queries and multi-attribute range queries over resource names. Our system utilizes a descriptive naming scheme to name resources and a multi-dimensional resource ID space for message routing through an overlay network of name servers (NSs). The resource ID space is constructed on the overlay network based on CAN routing algorithm. Our novel mapping scheme between resource names and resource IDs preserves resource ID locality while still achieving good load balancing regarding resource information distribution. We also propose a multicast routing algorithm to deliver resource information and a broadcast routing algorithm to route query messages to corresponding NSs with small cost of message transmission. Our simulation results show that our system can achieve good routing performance and load balancing.