• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.199-201
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• 2002

인터넷 이용자 수의 급격한 증가와 인터넷 서비스 보급의 대중화에 따라 인터넷에서 최고의 미래가치를 갖는 기술로 VoIP(Voice over IP)기술이 부각되고 있다. 현재 가장 활발하게 연구되고 있는 VoIP 기술로는 H.323과 SIP(Session Initiation Protocol)가 있다. SIP는 사용자간에 멀티미디어 세션을 생성, 수정, 해제하는 응용 계층의 시그널링 프로토콜로서, H.323에 비해 단순한 구조를 가지고 있고 새로운 부가 서비스를 추가하기가 쉽다는 장점이 있어서 앞으로 많은 사용이 예상된다. 비록 SIP가 차세대 네트워크 및 서비스에 이용될 전망이지만, H.323 기반의 기존 서비스를 무시할 수 없으므로 H.323 기반의 서비스를 사용자에게 투명하게 제공하면서 SIP 기반 구조로의 변환 모색이 필요하다 따라서, 본 논문에서는 SIP와 H.323간의 연동 서비스를 제공하는 SIP-H.323 Translator의 설계에 대해 기술하고자 한다.

• 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 Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.2
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• pp.575-578
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• 2005

인터넷 사용자의 급격한 증가로 인하여 인터넷 서비스 보급이 대중화됨에 따라 인터넷을 이용한 다양한 부가 서비스들이 창출되고 있다. 이러한 흐름에 발맞추어 VoIP(Voce over IP) 표준기술들이 최근 몇 년간 주요 이슈가 되고 있다. VoIP 서비스가 시장성 있는 기술로서 각광을 받게 되면서, VoIP 서비스를 위한 시스널링 프로토콜인 SIP(Session Initiation Protocol)가 기존의 H.323을 대체하는 기술로서 주목 받고 있다. 또한 SIP를 이용한 다양한 부가 서비스들도 함께 관심이 집중되고 있으며, 그중에서도 SIP를 이용한 프레즌스 서비스에 많은 관심을 보이고 있다. 본고에서는 현재 IETF를 중심으로 진행되고 있는 SIP 기반 프레즌스 부가 서비스에 대한 표준 기술 및 주요 이슈에 대해 소개하고자 한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2014.11a
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• pp.449-452
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• 2014

최근 이동통신 사업자들은 All-IP 기반의 서비스를 개발하고 상용화하기 위해 힘쓰고 있다. 그 이유는 All-IP 기반의 서비스가 LTE의 넓은 대역폭을 사용하여 기존 서비스와는 현저한 차별성을 가지고 있기 때문이다. 음성통화를 LTE 기반으로 제공하는 VoLTE 서비스도 그 중의 하나로서 현재 이동통신 3사 모두 상용화하여 이 새로운 고음질 및 고화질 커뮤니케이션 서비스에 대해 마케팅을 벌이고 있다. 하지만 VoLTE 서비스는 보안에 대한 충분한 고려가 이루어지지 않은 상태로 상용화되었으며, VoLTE에서 사용되는 SIP(Session Initiation Protocol) 프로토콜을 악용한 여러 유형의 공격에 매우 취약하다. 본 논문에서는 VoLTE 서비스에 대한 보안 위협 중 가장 기본이 되는 스캐닝 공격에 대해 기술하고 이를 탐지할 수 있는 방안을 제시한다.

• Journal of the Korea Computer Industry Society
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• v.5 no.9
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• pp.1017-1024
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• 2004

In this paper, we propose the progressed interworking method between H.323 and SlP, then explain the improved property. The VolP(Voice over Internet Protocol) technology which is able to use a voice service through internet is more cheaper then existing telephone charges, and is easil)· accept the various of multimedia services from internet. Previous connectionmethod of VoIP used H.323 protocol, but it is very complex to connection establishment. so, the SIP(Session Initiation Protocol) protocol that propose in SIP-Working Group is in use recently. Therefore, we need new interworking methodology between H.323 and SIP Products. In this thesis, the progress interworking method between H.323 and SIP are Propose, then interpret unnecessary packet delay for call setup and improved feature of message exchange.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.1
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• pp.32-39
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• 2015

The voice communications with wireless backbone based networks are evolving into a packet switching VoIP systems. In those networks, a call processing scheme is required for management of subscribers and connection between them. A VoIP service scheme for those systems requires reliable subscriber management and connection establishment schemes, but the conventional call processing schemes based on the centralized server has lack of reliability. Thus, the mutual-backup architecture of SIP-servers is required to ensure efficient subscriber management and reliable VoIP call processing capability, and the synchronization and call processing schemes should be changed as the architecture is changed. In this paper, a mutual-backup architecture of SIP-servers is proposed for wireless backbone based networks. A message format for synchronization and information exchange between SIP servers is also proposed in the paper. This paper also proposes a FSM scheme for the fast call processing in unreliable networks to detect multiple servers at a time. The performance analysis results show that the mutual backup server architecture increases the call processing success rates than conventional centralized server architecture. Also, the FSM scheme provides the smaller call processing times than conventional SIP, and the time is not increased although the number of SIP servers in the networks is increased.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.4
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• pp.678-683
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• 2003

This study presents a result of performance with the design of network topology for voice and data integration under computer network. This network is consisted of FastEthernet, other LANs and ATM WAN(wide area network), and performance evaluation of delay in a PBX+IP network, delay in a VoIP network and delay in a IP+ATM network will be shown. We use parameters including network bandwidth, number of packet, routing protocol(IGRP, OSPF). We simulate integrated of voice and data used PBX. we will study further about the case of integrated of voice and data environments using PBX. and, evaluate IP+ATM WAN average measured network delay and average delay of VoIP network.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.682-685
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• 2002

VoIP(Voice over IP) Technology is highlighted because of easy adopting the value added services related voice In this paper, we described the Internet telephony service based on SIP. Especially, we described the extension for re-INVITE function. Re-INVITE function is useful for cail transfer service or conference service.

• Journal of KIISE:Computing Practices and Letters
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• v.11 no.4
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• pp.312-333
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• 2005

SIP(Session Initiation Protocol) has been proposed for session control protocol of Internet multimedia communication system like VoIP(Voice over IP). SIP has complicated session control steps to support various kinds of audio and video formats and to assure service quality of real time data communication. Up until now, existing protocol analyzers can not provide such detailed information of SIP based communication system. In this paper, therefore, we propose a new protocol analyzer as a tool that can analyze and diagnose SIP based multimedia communication system throughout the session initiation, data exchange and session change steps. The propose traffic analyzer, which is called STAT(SIP based Traffic Analysis Tool), Is implemented on Winder's environment so that it is generally usable and extensible. Since STAT analyze low level packets captured via Ethernet broadcasting property, it is able to provide session status and real time traffic monitoring information without any affection to the communication system. The STAT which is implemented in this paper. therefore, is expected to be a useful tool for developing and managing of a SIP based multimedia communication system.

• Journal of Korean Institute of Industrial Engineers
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• v.32 no.1
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• pp.1-8
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• 2006

A modified PCF(Point Coordination Function) protocol with the optimal polling sequence is defined in detail and shown to improve the efficiency of the conventional PCF protocol in IEEE 802.11 wireless LAN standard. The problem for the optimal polling sequence is formulated as TSP(Travelling Salesman Problem) with the distance values of 1's or 0's. Numerical examples show that the optimal polling sequence increases the capacity of the real-time service such as VoIP(Voice over Internet Protocol).