• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.703-705
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• 2003

현재 인터넷 서비스의 근간을 형성하고 있는 IPv4의 가용 주소 공간의 고갈, 보안성의 결여, 그리고 멀티미디어 서비스를 위한 QoS(Quility of Service)의 필요성과 같은 요구사항을 바탕으로 차세대 인터넷 프로토콜(IPv6)로의 전환이 요구되고 있다. 본 연구 목적은 이러한 네트워크상의 이동 인터넷 환경에다 실시간 서비스를 제공할 수 있도록 SIP(Session Initiation Protocol)를 적용하여 통함 된 환경이 이전 보다 안전한 인터넷 정보서비스를 제공할 수 있도록 보안 메커니즘을 적용 하였다. 네트워크 계층과 응용 계층의 이동성 관리 모델의 통합은 전체적인 시그널링 부하를 줄이고 지속적인 통신을 위한 빠른 핸드오프를 제공한다. 즉, 본 연구는 현재 Mobile IPv6 에서 보안상 취약점으로 나타나는 문제점 및 SIP 보안 고려사항 및 이동성을 해결하기 위해 제안되는 해결방안들을 분석하고 적합한 보안 메커니즘 적용 방안을 제안 하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.556-559
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• 2001

VoIP(Voice over IP) Technology, turn voice services over traditional telephone network into internet, is highlighted because of easy adopting the value added services related voice In this paper, we described the user agent system architecture for internet telephony services based on SIP (Session Initiation Protocol)

• Proceedings of the Korea Information Processing Society Conference
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• 2008.05a
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• pp.698-701
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• 2008

본 연구는 멀티미디어 세션에 참여한 참여자간의 효율적인 의사소통과 상호협력 환경의 향상을 위하여 다양한 형태로 응용의 변화를 주는 세션 초기 프로토콜(SIP: Session Initiation Protocol)에서의 오류 관리기의 설계에 대한 연구이다. 제안하고자 하는 오류 관리기를 이용하면 멀티미디어 응용개발 환경에서의 오류 발생 시에 객체를 동적으로 생성 및 제거함으로써 자신의 컴퓨터 시스템 상황에 맞는 세션 초기 프로토콜에서의 세션을 진행할 수 있고, 유동적인 네트워크 트래픽에서도 진행 중인 세션 초기 프로토콜에서의 세션을 유지시킬 수 있을 뿐만 아니라, 오류가 발생된 응용을 제외한 객체만의 조합으로 다양한 형태의 세션 초기 프로토콜에서의 세션을 만들 수 있다.

• The Journal of the Korea Contents Association
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• v.4 no.1
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• pp.83-89
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• 2004

Instant messaging(IM) service is one of the useful communication means in the Intemet for exchanging simple messages between users. Usually, IM service is coupled with Presence service which provides status information of users. In this case, instant messages are sent and received directly between on-line users. Therefore, messages could not be exchanged when users are in off-line state. In this paper, We have implemented SIP based Instant Messaging System which includes message server that can store and manage instant messages. With this message server, messages could be exchanged legalness of user states. Furthermore, our system includes message waiting indication event package which provides useful informations about messages in the message server. And this system also includes the Caller Preference capabilities.

• Journal of Korea Multimedia Society
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• v.10 no.3
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• pp.335-347
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• 2007

VoIP (Voice over IP) is a technology to transport video and voice traffic over IP networks such as Internet. Today, the VoIP technology is viewed as the right choice for providing voice, video, and data communication among various terminals over the next generation network. This paper discusses a multipoint VoIP implementation with end-point mixing model which can support multipoint conference without a conference bridge. The multipoint VoIP is implemented with SIP (Session Initiation Protocol), and supports STUN (Simple Traversal of UDP Through NATs) since it works in an asymmetric NAT (Network Address Translator) environment. The characteristics of this paper are as follows. It is possible that all terminals in the hierarchical conference don't receive the duplicated media information because we use the end-point mixing model with the new media processing module. And, the paper solves the problem that the hierarchical conference session should be separated into several sessions when a mixing terminal terminates the hierarchical conference session.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.25 no.6
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• pp.1411-1420
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• 2015

Lawful interception(LI) standard of telephone networks has technical limitations to lawfully intercept IMS/SIP-based mobile communication network subscriber who using Android and iPhone device. In addition, the technical standards related to legal interception of the IMS/SIP of the wireless network is insufficient compared to the systematic study of the development of a wireless network infrastructure. The architecture proposed in the standard of ETSI(European Telecommunications Standards Institute) for the seamless LI is insufficient to overcome the limitations of traditional voice-centric LI techniques. This paper proposes an IMS/SIP-based architecture to perform LI under 3G networks that focuses on mobility-supported environments with merging cellular networks and the Internet. We implemented the simulation to verify the efficiency of the proposed architecture, and the experimental results show that our method achieves higher lawful interception rate than that of existing interception methods.

• Journal of KIISE:Computing Practices and Letters
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• v.13 no.5
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• pp.300-310
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• 2007

The Session Initiation Protocol (SIP) is one of the major protocols used in call-setup over IP telephony. The SIP-signaled calls use many-sided states according to a request of user. In this paper, we suggest seven states and some events that help developers to design and implement new applications efficiently. And they enable an object-oriented design of the system. If you design the call-setup procedure only by the processing model suggested in RFC 3261 over commercial network, a fatal error may occur in the system because of heavy data traffic or unpredicted exception cases. However, according to the suggested seven states, if they are predefined events in the current system state, the standardized processing routine is executed. Otherwise, they can be processed by the exception routine in system. All event processing routines are designed and implemented using Finite State Machine (FSM).

• The Journal of the Korea Contents Association
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• v.5 no.2
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• pp.257-266
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• 2005

As developing of the internet and computer technology, more interests are gathered to the conference service which provides capability of multi-party real-time visual conference. In this paper, we have designed and implemented a SIP-based visual conference system which includes Presence service. The elements of this conference system are user system, which has conference UA(User Agent) capability, presence seuer and conference server. For the presence service, we have adapted publication method which uses SIP PUBLISH message, and with this service various status informations of users are easily acquired. Also invitations and involvements to the conference are easily made through this service. For the conference server which controls establishment and management of multi-party connections, we have included conference event package. This package provides dynamically changing conference informations and users informations through SIP subscription and notification functions.

• The Journal of the Korea Contents Association
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• v.7 no.3
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• pp.17-24
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• 2007

SIP-based tightly coupled conference, which has a centrally located conference server for controlling and management, can be classified several models according to location of focus and mixer. These are centralized server model, endpoint server model, media server component model and distributed mixing model. However each model has its strength and weakness. In this paper, we propose and implement a SIP-based effective hybrid-type conference model which decreases amount of SIP signaling messages, lowers load of server media mixer, and can be easily expandable to large scale conference. In this model, when the number of participants exceeds a pre-defined limit, the conference server selects some participants which posses specific functions and let them share functions of notifications of conference state event package and media mixing. When each participant subscribes conference state event package to the server, it can indicates its possession of such functions by a specific header message. The server stores the indication to the conference information database, and later uses it to select participants for load sharing. The performance of our proposed model is evaluated by experiments.

• ETRI Journal
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• v.36 no.5
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• pp.865-875
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• 2014

Voice phishing (vishing) uses social engineering, based on people's trust in telephone services, to trick people into divulging financial data or transferring money to a scammer. In a vishing attack, a scammer often modifies the telephone number that appears on the victim's phone to mislead the victim into believing that the phone call is coming from a trusted source, since people typically judge a caller's legitimacy by the displayed phone number. We propose a system named iVisher for detecting a concealed incoming number (that is, caller ID) in Session Initiation Protocol-based Voice-over-Internet Protocol initiated phone calls. Our results demonstrate that iVisher is capable of detecting a concealed caller ID without significantly impacting upon the overall call setup time.