• Journal of Korean Institute of Industrial Engineers
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• v.24 no.2
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• pp.241-249
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• 1998

In this paper, we seek to find an optimal allocation of link capacity in a data communication network. The architecture of the data communication network considered in the study is an online-service network based on public switched telephone network(PSTN) and packet switched data network(PSDN). In designing the architecture of the network, we need to deal with various measures of quality of service(QoS). Two important service measures are the call blocking probability in PSTN and the data transfer delay time in PSDN. Considering the tradeoff between the call blocking probability and the data transfer delay time in the network, we have developed the optimal link capacity allocation model that minimizes the total link cost, while guarantees the call blocking probability and the data transfer delay time within an acceptable level of QoS. This problem can be formulated as a non-linear integer programming model. We have solved the problem with tabu search and simulated annealing methods. In addition, we have analyzed the sensitivity of the model and provided the insight of the model along with computational results.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.3
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• pp.652-663
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• 1996

In this paper, we analyze the blocking probability of a call in a PC communication network. In a PC communication network, a call generated from the PSTN (public switched telephone network) requests the connection to the ICPS(information communication processing system) which interconnects the the PSTN and the PSDN(packet switched data network). Then, as a second step, a packet call is requested to the specific service database attached in the PSDN. The call blocking probabilies are analyzed for these two interface points, the PSTN interface ports of the ICPS and the PSDN interface link of the database. We also present some numerical results to show the effects of the system parameters on the performance of the network. The results show the effects of the capacity of each interface point on the performance point.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.67-69
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• 2003

현재 인터넷은 음성을 포함한 실시간 정보의 제공을 기반으로 정보에 대한 욕구를 충족시키고 있으며, 이러한 인터넷의 실시간을 바탕으로 사용자는 새로운 서비스에 대한 요구를 창출하게 되었고, 저렴한 인터넷을 이용하여 Public Switched Telephone Network(PSTN)과 같은 기존 통신망을 대체하는 연구가 활발히 이루어지고 있다. VoIP(Voice over Internet Protocol)는 이러한 요구에 부흥하는 인터넷의 대표적인 서비스로 등장하고 있으며, MGCP, SIP 그리고 H.323 같은 프로토콜을 기반으로 VoIP 서비스를 위한 다각적인 접근과 연구가 진행 중이다. 본 연구는 VoIP 서비스를 위한 여러 프로토콜 중 IETF가 주관하고 있는 MGCP(Media Gateway Control Protocol) 스팩에 따라 MGCP를 구현하였으며, 댁내 서비스를 위한 인터넷에서의 VoIP 신뢰성을 보장하는 방안으로 기존 PSTN망을 백업형태로 지원하는 방안을 연구하여 특정 번호는 Call Agent(CA)와 MGCP 프로토콜로 통신하지 않고 임의 변경 없이 자동으로 기존 망으로의 발신이 가능한 시스템을 설계하고 구현하였다.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.16 no.12
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• pp.8357-8362
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• 2015

The emergency call device has been installed and operated for the elevator passenger safety, but the call can not be connected or delay to the operator in an emergency. The application of the Internet of Things(IoT) technology is required for the systematic management and reliability of the emergency call device installed in the elevator. To meet these requirements it is necessary to apply the RMS(Remote Management System). Emergency call device periodically reports the operating status information and the emergency call history information of the emergency call device to the RMS in the PSTN(Public Switched Telephone Network) environment, and transmits the call request information in an emergency. RAS(Remote Access Server) plays a role of converting the PSTN signals to TCP / IP. RMS reports a significant outbreak call and abnormal operating conditions to administrators. In this study, we developed optimized protocol to PSTN environment that is required by RMS and implemented its differentiated process. And a RAS-RMS Protocol was added to the Application Layer. Development protocol consisting of simple structure as compared to CoAP, MQTT or HTTP can work properly in low-spec CPU and transmit elevator emergency key information easily.

• International journal of advanced smart convergence
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• v.8 no.3
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• pp.69-77
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• 2019

Today, most of elevators have an emergency call facility for emergency situations. However, if the network installed in the elevator is also out of power, it cannot be used for the elevator remote monitoring and management. So, we develop an integrated and unified emergency call system, which can transmit not only telephone call but also data signals using PSTN(Public Switched Telephone Network) in order to remote monitoring and management of elevators, even though a power outage occurs. The proposed integrated emergency call system to process multiple data such as voice and operational information is a multi-channel board system which is composed of an emergency phone signal processing module and an operational information processing module in the control box of elevator. In addition, the RMS(remote management server) systems based on the Web consist of a dial-up server and a remote monitoring server where manages the elevator's operating information, status records, and operational faults received via the proposed integrated and unified emergency call system in real time. So even if there's a catastrophic emergency, the proposed RMS systems shall ensure and maintain the safety of passengers inside the elevator. Also, remote control of the elevator by this system should be more efficient and secure. In near future, all elevator emergency call system need to support multifunctional capabilities to transmit operational data as well as phone calls for the safety of passengers. In addition, for safer elevators, it is necessary to improve them more efficiently by combining them with high-tech technologies such as the Internet of Things and artificial intelligence.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.4
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• pp.744-751
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• 2006

Even many kinds of Internet-based services have been generated due to the great development of the Internet. PSTN still exists in the center of the national infrastructure network and the transit exchanges will be maintained the core roles in PSTN for many years in the future. These transit exchanges often suffer from unexpected overload situation because they have to process Intelligent Network calls and mobile calls additionally as well as POTS (Plain Old Telephone Service). In this paper, we suggest an efficient overload control algorithm for the distributed transit switching system under the various types of call services. We also show that this algorithm can be implemented easily cooperating with the network management control functions. The simulation technique is used to show that the proposed algorithm effectively controls calls and maintains safely the call processor's load under the various kinds of overload situations.

• Journal of Korean Society of Industrial and Systems Engineering
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• v.33 no.3
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• pp.137-145
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• 2010

The objective of this paper is to evaluate the equilibrium price between PSTN and VoIP telephony services in the case of non-linear utility function. Currently there are two types of wired phone services we are known PSTN (Public Switched Telephone Network) and VoIP (Voice over Internet Protocol). The PSTN telephony which provide high quality service and VoIP which provides relatively low quality service form a vertically differentiated oligopoly. Therefore, the evaluation of the equilibrium price between PSTN and VoIP services is very important to wired phone service providers. The equilibrium price depends on the state of the service cost function has been proved different value. This paper was evaluated each equilibrium price for the state of the linear cost function and non-linear cost function. Subsequently, this paper analyzed the demand of both services and the equilibrium profit which can maximize the profit of both service providers.

• The Transactions of the Korea Information Processing Society
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• v.6 no.10
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• pp.2740-2751
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• 1999

Critical to more widespread use of Internet telephony are the smooth interoperability with the existing telephone network and the improved quality of voice connections. Of these requirements, this interoperability comes through the use of Internet Telephony Gateway's which perform protocol translation between an IP network and the Public Switched Telephone Network. In this paper, we have focused on the necessity and possibility of interoperability, and furthermore derives the necessary requirements for interoperability between IP networks and PSTN. For this purpose, we have analyzed the signaling protocols for gateway system. Then, we have modelled the inter-working part using the Petri-Net model. Through reachability trees of the Petri-Net model, we have confirmed that interoperability is possible, and that characteristics of deadlock, liveness, and boundness are satisfied.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.406-408
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• 2002

미래의 통신망 구조인 NGN(Next Generation)으로의 이행을 위하여 현재의 방대한 PSTN(Public Switched Telephone Network) 인프라를 한번에 모두 교체할 수 없으므로 신설되는 VoIP(Voice on IP) 기반의 네트워크와 기존의 PSTN이 상호 연동되는 과도기적 과정으로 Pre NGN이 제안되고 있으나 SS7(Signaling System No 7)을 기반으로 하고 있어 아직도 상당수를 차지하는 SD(Space Division) 교환기의 교체는 불가피한 것으로 보인다. 본 논문에서는 SD 교환기를 TD(Time Division) 교환기로 교체하기보다는 미래의 NGN을 지향하며 소요비용의 절감을 위하여 Soft Switch기능을 보유한 AGW(Access Gate Way)로의 교체를 제안하고 현재의 망에서 필요한 기능 및 조건을 제시 하였다.

• The KIPS Transactions:PartC
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• v.12C no.2 s.98
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• pp.251-260
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• 2005

On the internet phone or PSTN(Public Switched Telephone Network), the automatic callback is an useful service in the case of busy state when one user calls the other. By using this service, automatic redial is possible when the other party hangs up. However, in the basic automatic callback service, the user who wants callback should wait until the other party hangs up even in the case of emergency. Therefore in this paper, to solve this problem we have extended CPL(Call Processing Language) and, within user system we have included and linked this extended CPL processing module and dialog event package which processes SIP INVITE initiated dialog state informations. We have implemented this system for being used in SIP(Session Initiation Protocol)-based VoIP(Voice over IP) system.