• ETRI Journal
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• v.25 no.1
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• pp.34-40
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• 2003

This paper considers a TMN-based management system for the management of public ATM switching networks using a four-level hierarchical structure consisting of one network management system, several element management systems, and several agent-ATM switch pairs. Using Jackson's queuing model, we analyze the effects of one TMN command on the performance of the component ATM switch in processing local calls. The TMN command considered is the permanent virtual call connection. We analyze four performance measures of ATM switches- utilization, mean queue length and mean waiting time for the processor directly interfacing with the subscriber lines and trunks, and the call setup delay of the ATM switch- and compare the results with those from Jackson's queuing model.

• Proceedings of the KIEE Conference
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• 1987.07b
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• pp.965-968
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• 1987

Conventional major electronic switching systems based on stored program control employ a Von Neumann styled control processor. It has strict limitations such that it essentially lacks concurrency in executing instructions, which have brought the software bottleneck problem, and the capabilities of call processing are restricted by expanding system's capacity. In this paper, a new architecture of call control processor based on the data flow system is proposed, aiming at fundamental resolution for these limitations. The processor has a number of advantages in such as expansibility of system's capacity, parallel processing of calls, and so on.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.1555-1558
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• 2002

SIP는 간단한 텍스트 기반의 응용계층 프로토콜로서, H.323을 대체할 수 있는 프로토콜이다. SIP는 인터넷 환경에 그대로 접목 할 수 있고, 새로운 기능 및 부가서비스의 제공이 용이하다는 장점을 가지고 있다. SIP 프로토콜은 요청메시지와 그에 대한 응답으로 구성되는 Request-Response방식이다. SIP의 장점은 유일한 개인 ID를 이용하여 장소와 단말기에 구애를 받지 않고 SIP서비스를 제공받을 수 있는 Personal Mobility Service에 있다. 본 논문에서는 User Agent와 프록시 서버(Proxy Server) 사이에 이루어지는 SIP 트랜잭션(Transaction)을 Call-Flow를 통해서 살펴보고자 한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2007.05a
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• pp.1527-1530
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• 2007

멀티미디어 디지털 기기의 발전은 Mobile Phone의 사용량 증가로 대두되고 있으며 그에 따른 문제를 해결하는 일도 시급한 상황이다. Mobile Phone 사용 중 작동이 멈춘다거나 하는 문제는 그 사용자수에 비례하여 발생 빈도나 위험성이 높아질 수밖에 없다. 이런 문제가 생기지 않도록 사전에 철저한 테스트를 해보고 시행착오를 겪어봐야 완벽한 제품으로 시중에 내놓을 수 있게 되는 것이다. 기존의 하드웨어와 연결해서 사용 하던 디버거와는 다르게 Remote Debugger는 이러한 문제점을 툴을 통해서 소프트웨어적으로 찾아내어 고치는 역할을 하게 될 것이다. 본 논문에서는 이러한 Remote Debugger를 구현하기 위한 부분 중 call stack watch view 에 대해 논의 하려고 한다.

• The KIPS Transactions:PartC
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• v.9C no.6
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• pp.927-934
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• 2002

In order to provide various multimedia services in a wireless network, the call admission control for wireless channels should be resolved at the time of call setup and handoff by moving mobile terminal. In this paper. we propose a reliable DCAC( Distributed Call Admission Control)scheme using virtual cluster concept. The proposed DCAC scheme considers the state of $1^{st}$ and $2^{nd}$ adjacent cells to provide a reliable call handling. The proposed scheme is analyzed by simulations and mathematical methods.

• The KIPS Transactions:PartC
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• v.11C no.4
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• pp.539-544
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• 2004

This paper presents the implementation of the bluetooth headset and the audio gateway connected to the mobile Phone in the embedded environment. The bluetooth module includes the BC02 processor chip, the BCSP02 firmware and the bluelab software Including bluetooth protocol stack. The above components in the bluetooth module developed at CSR company are used as the development environment. The application program using API functions supported by bluelab is coded by C language and loaded on the flash ROM of the bluetooth module. The cail processing capacity measuring the call setup time and the clearing time between the audio gateway and the headset is considered as the performance parameter of the developed systems. As a call setup and clearing time between the audio gateway and the headset is about 88.8ms, the call processing capacity is about 11 calls per second. Therefore the performance result is satisfied in the aspect of the call processing time.

• Journal of Information Processing Systems
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• v.2 no.2
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• pp.107-113
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• 2006

It is essential to guarantee a handoff dropping probability below a predetermined threshold for wireless mobile networks. Previous studies have proposed admission control policies for integrated voice/data traffic in wireless mobile networks. However, since QoS has been considered only in terms of CDP (Call Dropping Probability), the result has been a serious CBP (Call Blocking Probability) unfairness problem between voice and data traffic. In this paper, we suggest a new admission control policy that treats integrated voice and data traffic fairly while maintaining the CDP constraint. For underprivileged data traffic, which requires more bandwidth units than voice traffic, the packet is placed in a queue when there are no available resources in the base station, instead of being immediately rejected. Furthermore, we have adapted the biased coin method concept to adjust unfairness in terms of CBP. We analyzed the system model of a cell using both a two-dimensional continuous-time Markov chain and the Gauss-Seidel method. Numerical results demonstrate that our CAC (Call Admission Control) scheme successfully achieves CBP fairness for voice and data traffic.

• Journal of the Institute of Convergence Signal Processing
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• v.7 no.1
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• pp.24-32
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• 2006

In this paper, we proposed the multimedia call admission control algorithm with the bandwidth reservation based on the prediction of wireless terminal's location to guarantee quality of service for multimedia applications in cellular networks. This algorithm aims at minimizing possible errors In predicting the moving direction of terminals using a mobility prediction scheme. This prediction reduces the size of bandwidth reserved redundantly. In order to evaluate the performance of the algorithm, the blocking rate of new calls and the forced termination rate of hand-off calls are measured and compared the results with those of existing schemes. The results of the experiment revealed that the algorithm presented in this paper achieved better performance with lower call blocking rates and forced-termination rates than those of other methods.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.18 no.4
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• pp.1090-1100
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• 2024

While the incorporating ESG indicator is recognized as crucial for sustainability and increased firm value, inconsistent disclosure of ESG data and vague assessment standards have been key challenges. To address these issues, this study proposes an ambiguous text-based automated ESG rating strategy. Earnings Call Transcript data were classified as E, S, or G using the Refinitiv-Sustainable Leadership Monitor's over 450 metrics. The study employed advanced natural language processing techniques such as BERT, RoBERTa, ALBERT, FinBERT, and ELECTRA models to precisely classify ESG documents. In addition, the authors computed the average predicted probabilities for each label, providing a means to identify the relative significance of different ESG factors. The results of experiments demonstrated the capability of the proposed methodology in enhancing ESG assessment criteria established by various rating agencies and highlighted that companies primarily focus on governance factors. In other words, companies were making efforts to strengthen their governance framework. In conclusion, this framework enables sustainable and responsible business by providing insight into the ESG information contained in Earnings Call Transcript data.

• Journal of Korea Multimedia Society
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• v.12 no.2
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• pp.257-271
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• 2009

In this paper, we propose a protocol compression scheme for enhancing the performance of call processing of PTT(Push-to-Talk) which is one of the important services in IMS(IP Multimedia Subsystem), a next generation integrated wired/wireless packet communication network. To service the PTT on an IMS network, it should use the same call setup procedure as legacy Mobile and TRS(Trunked Radio System) networks and have a fast call setup time and enough communication bandwidth because a number of terminals should be able to exchange same data in real time. The proposed A+SigComp scheme reduces the initial call setup delay of SIP by about 10%, which is used by PTT service for call setup. In addition, the A+ROHC scheme is proposed to compress the header of RTP packets transferred during PTT voice transmission and, as a result, about 5% of increase in communication efficiency is observed.