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Design and Implementation of Internet Telephony Services  

이종화 (ETRI 표준연구센터 통신프로토콜표준연구팀)
강신각 (ETRI 표준연구센터 통신프로토콜표준연구팀)
Abstract
The fast advance in the VoIP technologies gives a rich opportunity to create different kind of VoIP applications such as IP telephony services. The application level call signaling protocols such as ITU-T H.323 and IETF SIP provide the communication functions of end-to-end call setup and release. Currently, there is a lot of H.323 based VoIP products in the market, however SIP is considered as a suitable protocol for supporting applications in IP environments, so SIP based VoIP products and services begin to appear. In this paper, firstly we present the characteristics of some possible SIP based applications and describe the design and implementation of a VoIP example service named PC-to-PC Internet telephony service using the developed SIP network components. The PC-to-PC Internet telephony service and User Agent are developed in MS window 98/2000 using visual C/C++, and Proxy server and Registrar in Linux 7.0 using C, respectively.
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Citations & Related Records
Times Cited By KSCI : 2  (Citation Analysis)
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[] / IETE RFC 1889; RTP: A Transport Protocol for Real-Time Applications
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[] / IETF RFC 2327; SDP:Session Description Protocol
3 SIP Telephony Call Flow Examples /
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4 SIP 기반 Stateless Proxy 서버와 Registrar 서버의 설계 및 구현 /
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[ 이종화;안상현 ] / 한국정보처리학회지   과학기술학회마을
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[] / Columbia University
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[] / IETF RFC 2543 ; SIP:Session Initation Protocol
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[] / Pingtel Home page