• The KIPS Transactions:PartC
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• v.11C no.6 s.95
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• pp.799-806
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• 2004

A growing diversity of pervasive devices is gaming access to the Internet and other information. However, much of the rich multimedia contents cannot be easily handled by the client devices because of the limited communication, processing, storage and display capabilities. The in-tegration of voice, data and video service modified the target of networking technologies. Networks must have some the capabilities for in-tegration of various services and also for QoS support as required by each of those service. Because of these reasons, we developed EAGIS(Efficient ATM Gateway for real time Internet Service) to provide seamless multimedia service between the ATM network and the Internet. EAGIS consists of the interworking unit, content server, transcoding server, and the serveice broker to provide seamless multimedia service be-tween the ATM network and the Internet. In this paper, we design the architecture and transcoding service scenario of the EAGIS. When the RTP is used for the bi-directional communication, transcoding time is configured by the time-stamp of RTCP. When HTTP is used for unidirec-tional communication, self-timer is used. By using these reference time, standard transcoding method is applicable according to the frame trans-mission rate and network traffic load. And we can also assure the QoS of the multiple users` effective bandwidth by our algorithm.

• Journal of Biomedical Engineering Research
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• v.23 no.3
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• pp.197-205
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• 2002

The objectives of this Paper is to implement a diagnostic classifier of differential laryngeal diseases from acoustic signals acquired in a noisy room. For this Purpose, the voice signals of the vowel /a/ were collected from Patients in a soundproof chamber and got mixed with noise. Then, the acoustic Parameters were analyzed, and hierarchical neural networks were applied to the data classification. The classifier had a structure of five-step hierarchical neural networks. The first neural network classified the group into normal and benign or malign laryngeal disease cases. The second network classified the group into normal or benign laryngeal disease cases The following network distinguished polyp. nodule. Palsy from the benign laryngeal cases. Glottic cancer cases were discriminated into T1, T2. T3, T4 by the fourth and fifth networks All the neural networks were based on multilayer perceptron model which classified non-linear Patterns effectively and learned by an error back-propagation algorithm. We chose some acoustic Parameters for classification by investigating the distribution of laryngeal diseases and Pilot classification results of those Parameters derived from MDVP. The classifier was tested by using the chosen parameters to find the optimum ones. Then the networks were improved by including such Pre-Processing steps as linear and z-score transformation. Results showed that 90% of T1, 100% of T2-4 were correctly distinguished. On the other hand. 88.23% of vocal Polyps, 100% of normal cases. vocal nodules. and vocal cord Paralysis were classified from the data collected in a noisy room.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.24 no.4
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• pp.625-634
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• 2014

Tactical networks is being operated in configuration that consisting of a variety of characteristics communication equipments and heterogeneous networks. In this configurations, end-to-end communication can be achieved using interworking gateway for converting the data format of the network and using encryption algorithm of the networks. The use of mechanism results in a problem that secure data cannot be transferred directly, reprocessing and processing delay of communication in heterogeneous tactical networks. That is, for encoding and decoding of data, the decryption of encrypted data and re-encryption processing must be required at the gateway between different networks. In this paper proposes to mechanism for end-to-end secure communication in heterogeneous tactical networks. Using the proposed method, end-to-end secure communication between heterogeneous tactical networks(PSTN-UHF networks) which removes the necessity of a gateway for converting data into data formats suitable for network to remove a transmission delay factor and enable real-time voice and data communication and achieve end-to-end security for heterogeneous tactical networks. we propose a novel mechanism for end-to-end secure communication over PSTN and UHF networks and evaluate against the performance of conventional mechanism. Our proposal is confirmed removal of security vulnerabilities, end-to-end secure communication in heterogeneous tactical networks.

• The KIPS Transactions:PartC
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• v.9C no.6
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• pp.935-944
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• 2002

It is the Mobile Ad Hoc Networks that constituted with serveral mobile node that can communicate with other mobile nodes. Until now, there were no routing protocols considering such as Multimediadata, VOD (Video On Demand), which is required of lots of bandwidth in Mobile Ad Hoc Network, io we are in the need of QoS (Quality of Service)routing protocol to transmit the data packets faster and more accurate. Also, there are an unidirectional links due to asymmetric property of mobile terminals or current wireless environments on practical mobile ad hoc networks. However, at present, the existing mobile ad hoc routing protocols are implemented to support only bidirectional links. In this paper, we propose the Advanced Routing routing protocol in order to implement a new routing protocol, which is fit to mobile ad hoc networks containing unidirectional links and to support QoS service. For the performance evaluation, we use NS-2 simulator of U.C. Berkeley. We could get not only increased received data rate and decreased average route discovery time, but also network load decreases with compared Best effort service.

• Journal of the Korean Association of Geographic Information Studies
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• v.18 no.1
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• pp.64-73
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• 2015

Recently, the integrated utilization of technology with IT is in demand for the effectiveness of field management together with the prevention and prompt action on safety accident at construction site. In addition, the establishment of construction site support system is necessary to implement the securing of worker's safety, smooth work instruction, efficiency in construction, and others. Data standardization and network architecture were designed regarding data and sound information for data transmission between systems and management. These were to construct USFSS based on integrated VI-GNSS technology in this research. In the stability test of data for each system constructed through it, around 98% stability was secured between workers and for transfer vehicle system within underground structure and field server system in regards to the data transmission stability, around 100% stability was secured between field server system and control system, respectively. Also, in the sound transmission stability test, around 99% reliability could be secured with 1km distance as its standard in case of sound transmission from underground structure construction site to field office near the field through wireless FRS system.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.2
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• pp.293-300
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• 2007

As the development of the various IP-based services, it is expected that Internet telephony service will gradually replace the traditional PSTN-based telephony service. But there are many issues resolved to spread the Internet telephony service. One of them is to support the emergency services in the Internet telephony. In the case of USA, it has been regulated that 911 services should be supported in the Internet telephony services using VoIP on the similar performance level to PSTN 911 service. According to the regulation, basic VoIP 911 calls should be routed to the general access line of LEA without the location information or the callback number, but the enhanced VoIP 911 calls with the location information and callback number should be routed on the dedicated 911 network and destined to the local 911 distribution center such as PSAP. But, in the current VoIP-based Internet telephony network, the emergency call service has not been handled as one of the special services as well at has a worse performance in comparison to it on PSTN. Moreover, the service has a critical problem that it can not be destined to the nearest PSAP because of the insufficient information about the location information and the call back number. In this paper, we suggest the SIP-based emergency call service mechanism in order to resolve the problems above mentioned. This suggested mechanism is implemented to show its effectiveness and efficiency.

• Journal of Intelligence and Information Systems
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• v.26 no.4
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• pp.87-109
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• 2020

Currently, thanks to the major stride made in developing wired and wireless communication technology, a variety of IT services are available on land. This trend is leading to an increasing demand for IT services to vessels on the water as well. And it is expected that the request for various IT services such as two-way digital data transmission, Web, APP, etc. is on the rise to the extent that they are available on land. However, while a high-speed information communication network is easily accessible on land because it is based upon a fixed infrastructure like an AP and a base station, it is not the case on the water. As a result, a radio communication network-based voice communication service is usually used at sea. To solve this problem, an additional frequency for digital data exchange was allocated, and a ship ad-hoc network (SANET) was proposed that can be utilized by using this frequency. Instead of satellite communication that costs a lot in installation and usage, SANET was developed to provide various IT services to ships based on IP in the sea. Connectivity between land base stations and ships is important in the SANET. To have this connection, a ship must be a member of the network with its IP address assigned. This paper proposes a SANET-CC protocol that allows ships to be assigned their own IP address. SANET-CC propagates several non-overlapping IP addresses through the entire network from land base stations to ships in the form of the tree. Ships allocate their own IP addresses through the exchange of simple requests and response messages with land base stations or M-ships that can allocate IP addresses. Therefore, SANET-CC can eliminate the IP collision prevention (Duplicate Address Detection) process and the process of network separation or integration caused by the movement of the ship. Various simulations were performed to verify the applicability of this protocol to SANET. The outcome of such simulations shows us the following. First, using SANET-CC, about 91% of the ships in the network were able to receive IP addresses under any circumstances. It is 6% higher than the existing studies. And it suggests that if variables are adjusted to each port's environment, it may show further improved results. Second, this work shows us that it takes all vessels an average of 10 seconds to receive IP addresses regardless of conditions. It represents a 50% decrease in time compared to the average of 20 seconds in the previous study. Also Besides, taking it into account that when existing studies were on 50 to 200 vessels, this study on 100 to 400 vessels, the efficiency can be much higher. Third, existing studies have not been able to derive optimal values according to variables. This is because it does not have a consistent pattern depending on the variable. This means that optimal variables values cannot be set for each port under diverse environments. This paper, however, shows us that the result values from the variables exhibit a consistent pattern. This is significant in that it can be applied to each port by adjusting the variable values. It was also confirmed that regardless of the number of ships, the IP allocation ratio was the most efficient at about 96 percent if the waiting time after the IP request was 75ms, and that the tree structure could maintain a stable network configuration when the number of IPs was over 30000. Fourth, this study can be used to design a network for supporting intelligent maritime control systems and services offshore, instead of satellite communication. And if LTE-M is set up, it is possible to use it for various intelligent services.

• Asia pacific journal of information systems
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• v.17 no.2
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• pp.97-122
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• 2007

Conventional networking and telecommunications infrastructure characterized by wires, fixed location, and inflexibility is giving way to mobile technologies. Numerous research reports point to the ultimate domination of wireless communication. With the increasing prevalence of advanced cell-phones, various mobile data services (hereafter MDS) are gaining popularity. Although cellular networks were originally introduced for voice communications, statistics indicate that data services are replacing the matured voice service as the growth engine for telecom service providers. For example, SK Telecom, the Korea's largest mobile service provider, reported that 25.6% of revenue and 28.5% of profit came from MDS in 2006 and the share is growing. Statistics also indicate that, in 2006, the average revenue per user (ARPU) for voice didn't change but MDS grew seven percents from the previous year, further highlighting its growth potential. MDS is defined "as an assortment of digital data services that can be accessed using a mobile device over a wide geographic area." A variety of MDS have been deployed, with a few reaching the status of killer applications. Many of them need to access the Internet through the cellular-phone infrastructure. In the past, when the cellular network didn't have acceptable bandwidth for data services, SMS (short messaging service) dominated MDS. Now, Internet-ready, next-generation cell-phones are driving rich digital data services into the fabric of everyday life, These include news on various topics, Internet search, mapping and location-based information, mobile banking and gaming, downloading (i.e., screen savers), multimedia streaming, and various communication services (i.e., email, short messaging, messenger, and chaffing). The huge economic stake MDS has on its stakeholders warrants focused research to understand associated dynamics behind its adoption. Lyytinen and Yoo(2002) pointed out the limitation of traditional adoption models in explaining the rapid diffusion of innovations such as P2P or mobile services. Also, despite the increasing popularity of MDS, unexpected drop in its usage is observed among some people. Intrigued by these observations, an exploratory study was conducted to examine decision factors of MDS usage. Data analysis revealed that the increase and decrease of MDS use was influenced by different forces. The findings of the exploratory study triggered our confirmatory research effort to validate the uni-directionality of studied factors in affecting MDS usage. This differs from extant studies of IS/IT adoption that are largely grounded on the assumption of bi-directionality of explanatory variables in determining the level of dependent variables (i.e., user satisfaction, service usage). The research goal is, therefore, to examine if increase and decrease in the usage of MDS are explained by two separate groups of variables pertaining to information quality and system quality. For this, we investigate following research questions: (1) Does the information quality of MDS increase service usage?; (2) Does the system quality of MDS decrease service usage?; and (3) Does user motivation for subscribing MDS moderate the effect information and system quality have on service usage? The research questions and subsequent analysis are grounded on the two factor theory pioneered by Hertzberg et al(1959). To answer the research questions, in the first, an exploratory study based on 378 survey responses was conducted to learn about important decision factors of MDS usage. It revealed discrepancy between the influencing forces of usage increase and those of usage decrease. Based on the findings from the exploratory study and the two-factor theory, we postulated information quality as the motivator and system quality as the de-motivator (or hygiene) of MDS. Then, a confirmative study was undertaken on their respective role in encouraging and discouraging the usage of mobile data service.

• The Transactions of the Korea Information Processing Society
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• v.5 no.11
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• pp.2954-2972
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• 1998

It is expected that the transport of compressed video will become a significant part of total network traffic because of the widespread introduction of multimedial services such as VOD(video on demand). Accordingly, VBR(variable bit-rate) encoded video will be widely used, due to its advantages in statistical multiplexing gain and consistent vido quality. Since the transport of video traffic requires larger bandwidth than that of voice and data, the characterization of video source and traffic modeling is very important for the design of proper resource allocation scheme in ATM networks. Suitable statistical source models are also required to analyze performance metrics such as packet loss, delay and jitter. In this paper, we analyzed and described on the characterization and traffic modeling of MPEG video sources. The models are broadly classified into two categories; i.e., statistical models and deterministic models. In statistical models, the models are categorized into five groups: AR(autoregressive), Markov, composite Marko and AR, TES, and selfsimilar models. In deterministic models, the models are categorized into $({\sigma},\;{\rho}$, parameterized model, D-BIND, and Empirical Envelopes models. Each model was analyzed for its characteristics along with corresponding advantages and shortcomings, and we made comparisons on the complexity of each model.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.6
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• pp.137-145
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• 2010

Digital vessel have to accurate and efficient mange the digital data from various sensors in the digital vessel. But, In sensor network, it is difficult to transmit and analyze the entire stream data depending on limited networks, power and processor. Therefore it is suitable to use alternative stream data processing after classifying the continuous stream data. In this paper, We propose efficient processing method that arrange some sensors (temperature, humidity, lighting, voice) and process query based on sliding window for efficient input stream and pre-clustering using multiple Support Vector Machine(SVM) algorithm and manage hash table to summarized information. Processing performance improve as store and search and memory using hash table and usage reduced so maintain hash table in memory. We obtained to efficient result that accuracy rate and processing performance of proposal method using 35,912 data sets.