The present paper proposes a dynamic spectrum access policy for multi-hop cognitive radio networks (CRNs), where the transmission in each hop suffers a delay waiting for the communication channel to become available. Recognizing the energy constraints, we assume that each secondary user (SU) in the network is powered by a battery with finite initial energy. We develop an energy-efficient policy for CRNs using the Markov decision process, which searches for spectrum opportunities without a common communication channel and assigns each sensor's decision to every time slot. We first consider a single-sensor scenario. Due to the intermittent activation of the sensor, achieving the optimal sensing schedule requires excessive complexity and is computationally intractable, owing to the fact that the state space of the Markov decision process evolves exponentially with time variance. In order to overcome this difficulty, we propose a state-reduced suboptimal policy by relaxing the constrained state space, i.e., assuming that the electrical energy of a node is infinite, because this state-reduced suboptimal approach can substantially reduce the complexity of decision-making for CRNs. We then analyze the performance of the proposed policy and compare it with the optimal solution. Furthermore, we verify the performance of this spectrum access policy under real conditions in which the electrical energy of a node is finite. The proposed spectrum access policy uses the dynamic information of each channel. We prove that this schedule is a good approximation for the true optimal schedule, which is impractical to obtain. According to our theoretical analysis, the proposed policy has less complexity but comparable performance. It is proved that when the operating time of the CRN is sufficiently long, the data reception rate on the sink node side will converge to the optimal rate with probability 1. Based on the results for the single-sensor scenario, the proposed schedule is extended to a multi-hop CRN. The proposed schedule can achieve synchronization between transmitter and receiver without relying on a common control channel, and also has near-optimal performance. The performance of the proposed spectrum access policy is confirmed through simulation.

A new frame-based packet scheduling discipline designed for variable length packets is discussed. This algorithm extends a weighted round robin scheduling so that it can serve each queue fairly even if the length of packets varies. It can also handle burstiness by one of its important properties which we call "elasticity." In this paper, an overview of the proposed algorithm is described, and then it is systematically analyzed in terms of several important performance parameters: throughput, work complexity and elasticity. The proposed algorithm is evaluated by simulation and compared with the normal weighted round robin with regard to throughput and delay performance, thereby validating fairness of the algorithm with various types of traffic. Focusing on the complexity of the scheduling algorithm, we further propose and evaluate an improved version of the proposed algorithm, which provides the same properties with lower work complexity.

Recently, because of the expansion of TCP/IP networks, the evaluation of IP networks is being considered as a major task for network operators and administrators. As a result, many IP evaluation tools have appeared, but almost all of them can only be applied to a limited purpose. The authors have already developed an IP evaluation tool, called KITS (KDD Internet Test System) that can generate flexible load traffic. Based on KITS, the authors have now developed a new IP measurement tool. This paper describes the flexible and general IP evaluation tool, STAGE (Simulated Traffic Analyzer/generator for General Evaluation).

Today, many kinds of real-time applications are available over IP networks. It is important to measure the network performance for such applications before making use of real applications. Authors developed the general purpose traffic measurement tool for IP networks. This system can generate any kinds of traffic flexibly and calculate the network performance such as throughput, delay and loss ratio according to received packets. In this paper, the concept of this traffic measurement tool is described in detail, and several examples of network measurements are shown.

Effective communication strategies with a properly designed source precoding matrix (PM) and a properly designed relay beamforming matrix (BM) can significantly improve the spectral efficiency of multiple-input multiple-output (MIMO) relaying broadcast channels (RBCs). In the present paper, we first propose a general communication scheme with non-regenerative relay that can overcome the half-duplex relay constraint of the general MIMO-RBC. Based on the proposed scheme, the robust source PM and relay BM are designed for imperfect channel state information at the transmitter (CSIT). In contrast to the conventional non-regenerative relaying communication scheme for the MIMO-RBC, in the proposed scheme, the source can send information continuously to the relay and users during two phases. Furthermore, in conjunction with the advanced precoding strategy, the proposed scheme can achieve a full-degree-of-freedom (DoF) MIMO-RBC with that each entry in the related channel matrix is considered to an i.i.d. complex Gaussian variable. The robust source PM and relay BM designs were investigated based on both throughput and fairness criteria with imperfect CSIT. However, solving the problems associated with throughput and fairness criteria for the robust source PM and relay BM designs is computationally intractable because these criteria are non-linear and non-convex. In order to address these difficulties, we first set up equivalent optimization problems based on a tight lower bound of the achievable rate. We then decompose the equivalent throughput problem into several decoupled subproblems with tractable solutions. Finally, we obtain the suboptimal solution for the throughput problem by an alternating optimization approach. We solve the fairness problem by introducing an adjusted algorithm according to the throughput problem. Finally, we demonstrate that, in both cases of throughput and fairness criteria, the proposed relaying communication scheme with precoding algorithms outperforms existing methods.

Since Vendler classified aspect into four categories, state, achievement, activity, and accomplishment, much effort has been made to define the notion of aspect logically. It is commonly agreed that aspect represents the general temporal characteristics of events and states. However, there still remains a considerable amount of disagreement about its formal treatment. One of the major problems is that the aspect of a sentence shifts by certain types of sentence construction. For instance, adding time adverbials to a sentence modifies the original aspect, taking the progressive form of the verb changes the aspect, and so on. These phenomena are known as the aspect shifts. The other is the problem known as the imperfective paradox. The imperfective paradox is a problem of the truth definition of the progressives. The truth condition of the progressive form of the sentence is defined at an internal subinterval of the temporal range of the corresponding non-progressive sentence. If the truth condition of the progressive form of the sentence is defined using the truth condition of the non-progressive form of the sentence, there are logical contradictions of truth definition in a sentence such as "Max was building a house, but he never built it". These problems cause much confusion (1) in the truth definition of aspects, (2) in the definition of aspect operations, such as initiative, terminative, progressive, perfective, etc., and also (3) in the definition of adding time adverbials. This paper reviews the semantic problems with respect to aspect, and presents a consistent mechanism of aspect interpretation in order to settle all these semantic puzzles at once. For the sake of logical clarity, we construct a formal language, Lt, where every meaningful formula is a pair of a meaningful sentence and its aspect. The syntax of Lt describes the phenomenology of aspect shifts. The semantics of Lt defines temporal interpretation for all the meaningful sentences of Lt, with assuming the temporal interpretations of three inherent aspects, state, achievement, and activity. The proposed aspect interpretation gives a reasonable account for aspect shifts, and solves the imperfective paradox by asssuming the time structure to be backwards linear.

As the number of Internet applications in critical situations increases, the quality assurance of the network infrastructure becomes more and more important. The operators of such networks have to maintain the network as reliable and available with sufficient performance. Measurement technology is the key to maintaining the network condition and provides the network as being the important infrastructure of the network society. To meet these requirements, a high-speed and accurate IP meter, HIM, has been developed. Specific hardware has been made to realize the necessary specifications that enable data capturing with 20-µsecond order timing information from a gigabit class network. The experimental results with HIM show that: 1) software-based equipment cannot handle 100-µsecond order timing information, 2) the current computer system can transmit a high-speed (30 Mpbs) multimedia stream in accurate timing (jitter less than 20 µseconds) with UDP, 3) but TCP's retransmission mechanism makes the jitter worse (up to 264 mil seconds) and, 4) WAN's one-way traffic exhibits some strange behavior, which requires further investigation.

An extended version of weighted round robin scheduling algorithm for variable length packets is discussed. This can serve each flow fairly even if the length of packets varies. Further, an adaptive scheduling algorithm for bursty traffic is added to better treat multimedia sources. In this paper, an overview of the algorithm is described. Subsequently the throughput, latency, work complexity and elasticity for the algorithm are systematically defined and analyzed. These results show advantages of the algorithm compared with a normal or packet-based weighted round robin algorithm.

In this paper, we propose an SNMP-aware web cache design that has two main objectives: (1) to avoid overload of network devices by SNMP requests, and (2) guaranteeing the monitoring time granularity of SNMP Object Identifiers (OID) for a large scale network such as the Internet. To meet these objectives, a cache is built into an RESTful active proxy, called Tambourine, which is the gateway for accessing management information through the Internet. Tambourine changes the landscape of traditional SNMP monitoring by allowing the Internet users to monitor closed-domain network devices through translating requests in HTTP into SNMP. However, the typical web cache algorithm can not be used in Tambourine due to two main reasons: (1) SNMP is not a cache-aware protocol and therefore can not provide Tambourine with the caching rules that need to be applied, and (2) the cache in Tambourine needs to accommodate two SNMP monitoring patterns: periodic and on-demand polling. In order for efficient periodic polling, SNMP traffic is reduced by a multi-TTL cache and user (or Manager)-side aggregation. For efficient on-demand polling, four-state transition is used to categorize OIDs into dynamic and static objects, each of which is allocated an optimum TTL. To provide users with a proper time stamp, the cache time stamp is included in the response to the users' request. Our experiments show that our cache design gives the staleness of 0 and a bounded number of SNMP requests even when the number of users' requests goes to infinity.

This paper looks at the history of research in the Technical Committee on Information Networks from the time of its inception to the present and provides an overview of the latest research in this area based on the topics discussed in recent meetings of the committee. It also presents possible future developments in the field of information networks.

This paper presents the basic design and principles of an FQDN-based Internet Architecture, where a host is identified in the Internet only by its FQDN (Fully Qualified Domain Name). The address shortage problem for the IPv4 paradigm has been marginally solved by introducing private addresses within the intranet and also by DHCP for public Internet access services. These two approaches have been independently developed. Considering more elaborate peer-to-peer communications such as voice over IP (VoIP) in the future Internet, the shortage of address space will become a serious problem for a call or session to be established. To alleviate this, this paper proposes an FQDN-based Internet architecture, assuming that a global endpoint identifier (EID) in the network is FQDN instead of an IP address. Since a countably infinite number of addresses can be logically produced by FQDN, this system resolves the address shortage problem assuming a caller uses the FQDN of the called terminal or host. This requires a dynamic address loading system from the FQDN of the called terminal into one of the available public IP addresses. After proposing a possible address loading system, some mathematical results on the required number of public addresses for VoIP traffic, the address space size of the derived Internet, etc. are also presented.

As TCP/IP networks develop, various type of applications or services are appearing. Especially, many people want to use real time multicast applications over TCP/IP networks like a TV conference system. Most of the current TCP/IP networks, however, still do not support QoS guarantees using RSVP, so that they provide only a best-effort service. Therefore, such real time applications must control data transmitting rate by the network or receiver's condition. However, it is difficult to control data rate over a multicast session, since every receiver on a multicast network does not necessarily have the same environment. To solve this problem, the authors proposed a locally adaptive control intermediate system. This paper describes a rate-adaptive real-time multicast system with locally adaptive packet flow control.