• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4A
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• pp.377-384
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• 2006

According to fast VoIP technology development, more and more companies change voice network into IP based network among branch offices. IP PBX, which is deployed up to now, composed of IP phone and VoIP Gateway. Every telphone has replaced with If phone which support VoIP and VoIP gateway is installed in PBTN connection point to relay voice data. It can reduce the communication expense of International call, long distance call and call between a headquater and a trance because it uses internet line. In this paper, IP PBX is implemented that can distribute call using PBX network only usig personal terminal without Proxy Server. Depending on Role, terminal can be registered Master, Server and Client and it is verified in terms of performance and validation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.1
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• pp.134-144
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• 2009

The Ubiquitous/Unified Communication Platform supports various multimedia communication tools such as VoIP, Email, Unified Messaging, Instant Messaging, Web Conferencing, Audio/Video Communication etc. In this paper we introduced the main functions and architecture of the Unified Communication Platform and we researched on the function analysis and design of the VoIP System that supports PC-to-PBX/PSTN Phone and PBX/PSTN Phone-to-PC communications through the connectivity and interoperation with PSTN.

• The Journal of the Korea Contents Association
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• v.6 no.8
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• pp.20-29
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• 2006

This paper proposes a quality measurement model for video phone service over IP environment. Proposed model enhances conventional E-Model by using quality analysis and this model is suitable for ubiquitous environment. This research measures video phone quality by applying bust packet loss and recency effect. It uses delay and recency effect for compensating actual quality and recognized quality of user using NR and UR factor. Simulation results show that this model can provide more precise results than conventional model by considering recency effect of video phone service quality measurement model.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.10B
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• pp.676-688
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• 2005

We introduced mathematical economic analysis model for understanding the fairness of charge from VoIP providers for interconnection of access network. In order to set up this model we made four assumptions predictable in the real world. Also we proposed two accounting method that is flat-rate-pricing and usage-based-pricing and tried to propose which method is more desirable to charge for interconnection on the basis of social welfare and activation of market competitiveness. The outcome of this study includes the reasonable accounting method for interconnection between telephone network and IP Phone network which is most effective to ensure the social welfare and market competitiveness

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.11-14
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• 2005

In this paper, we propose a method for transmitting SMS(Short Message Service) for VoIP(Voice over IP) service supporting multi-protocol. The multi-protocol VoIP under consideration are generally composed of H.323, SIP and MGCP and Most ITSPs(Internet Telephony Service Provider) provide VoIP service with H.323 and SIP now. SMS is killer application in mobile telecom service and many people of various field use that service. For example, user can send many SMS messages and substitute e-mail. Also SMS can be provided with various internet application. Ahn, legacy phone of KT, can use SMS. Therefore VoIP phone also can be required with the same requirement. With the multi-protocol VoIP we will propose several methods sending efficiently SMS. To show the validity of the proposed method some examples are given in which the results can be expected by intuitive observation.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.105-108
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• 2000

CTI[1]의 응용 영역 중에서 인터넷 폰이 최근 뜨거운 관심의 대상으로 떠오르고 있다. 인터넷을 이용한 음성전달 기술은 인터넷의 성장 보급과 더불어 나날이발전 하고 있는데, 이러한 음성전달기술을 이용해 개발된 소프트 웨어를 통칭해서 인터넷 폰이라고 부르고 있다. 이러한 변화 속에서 비용의 절감과 비디오 전화, 영상회의와 같은 응용에 적용할 수 있는, 본 개발은 One Encoder One Decoder 지원의 VoIP(Voice over Internet Protocol) Phone에 관한 것으로, 특히 압축하여 인터넷 망에 접속시켜 사용할 수 있는 PC 장착형 One Board 형태의 시스템을 구현하였다. 이 Board에 사용 된 칩셋은 국내 회사인 C&S Technology 사의SEAGUL723이며, PC인터페이스는 PCI(Peripheral Component Interconnect) 버스 방식을 이용하였다. 주요 연구내용에 있어서 하드웨어 부분은 내선제어모듈,PCI 모듈, 칩셋을 이용한 음성신호처리 모듈, Board제어 모듈 등이 있으며, 소프트 웨어 설계 부분에 있어서는 하드웨어 구동을 위한 시스템 드라이브, Application과 인터넷 상의 VoIP 통신을 위한 소프트웨어, 사용자를 위한 User Interface 소프트 웨어 등이 있다.

• Journal of Korea Society of Digital Industry and Information Management
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• v.7 no.2
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• pp.39-46
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• 2011

Privacy IP hacking problems such as invasion of privacy, password cracking, voice wiretapping and internet over charged occurred, because VoIP internet voice phone service gradually spread. This thesis attempted to attack the VoIP service network by application. First use application to spoof IP address then attempted wiretap the VoIP service and sends a lot of messages to disturb service movement. At this point, we connected VoIP soft terminal, so we can operate real-time filtering operator to block the SIP Flooding offence by monitor the traffic and detect the location where it got attacked. This thesis used experiment to prove it is possible to detect the offence and defend from SIP Flooding offence.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.10a
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• pp.153-155
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• 2010

Internet phone (VoIP) services in Korea have achieved noticeable growth year after year since the service launching, and the growth still coninues. The market of mobile internet phone also expands sharply. Therefore, it is crucial to deploy networks which can support mobile internet phone services with excellent quality. For mobile internet phone services, it will be necessary to build and use networks with good mobility and high transmission rate. Current wireless networks for Internet services include 3G, Wi-Fi, and mobile WiMAX networks. 3G provides good mobility but lower transmission rate, whereas Wi-Fi exhibits excellent transmission rate but less mobility. Mobile WiMAX networks taking the merits of both, high mobility and transmission rate, are being deployed widely in recent years. This article examines the recent standardization efforts of WiMAX Forum for VoIP service in WiMAX networks.

• 한국정보통신설비학회:학술대회논문집
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• 2009.08a
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• pp.99-103
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• 2009

Recently, as wired and wireless communication services have rapidly developed and multimodal mobile devices which have various characteristics have widely spread, the need for new convergence services increases. The growing population of VoIP technologies and the high communication expense yield that the market of IP based telephony such as WiFi phone and IP phone is substituted for one of the conventional PSTN telephony. With the help of this trend, the wireline network operators desire to find a market in mobile networks. Therefore, they focus on Fixed Mobile Convergence (FMC) service as one of the key factors to accomplish this goal. FMC services are able to provide the mobility of voice services between circuit switched and packet switched networks. IP Multimedia Subsystem (IMS) based Voice Call Continuity (VCC) is one of the schemes to embody FMC services. As Application Server (AS) which has this VCC function provides seamless handover of services between heterogeneous networks, FMC subscribers can communicate seamlessly with others m WiFi domain and COMA domain using WiFi-COMA dual phone. Most of enterprises have already introduced IP network infrastructure and IP-PBX (Private Branch eXchange) for telephony. However, the problems of high communication cost and work inefficiency due to frequent outside jobs or business trips have remained. In order to solve these problems, demands for enterprise FMC services increase. In this paper, we introduce a new IP-PBX based VCC model that can provide seamless handover of voice services between WiFi and COMA networks for enterprise users and we investigate some interworking and security issues between Soft Switch (SSW) and IMS, or between IMSs. In addition, we introduce a new service that can provide the continuity of voice sessions as well as video sessions using Multimedia Session Continuity (MMSC) technology which has evolved from VCC. This service is expected to be one of the next-generation personalized services based on user's context.

• Journal of Korea Society of Digital Industry and Information Management
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• v.8 no.4
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• pp.151-157
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• 2012

This paper implements an effective attendance management and monitoring system utilizing the RFID(Radio Frequency IDentification) and Smart phone for student. In this paper, RFID system use 13.56MHz to control the attendance, and we can show that information of RFID and dynamic image through IP camera in Smart phone. We also make up the contents and database utilizing the labview program in computer. Especially, proposed technique can obtain dynamic image by server implementation at real time basis so that it is possible to use simultaneously several Smart phone. From the various and practical experiment, it is confirm that proposed system is useful for attendance management and remotely monitoring.