• Electronics and Telecommunications Trends
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• v.35 no.6
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• pp.68-77
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• 2020

Along the advancement of mobile networks, fixed telephone networks are gradually advancing from legacy networks based on copper and circuit-switches to optical cables and packet-switched IP networks. Incumbent fixed-line telephone operators are facilitating the introduction of IP networks and are gradually converting to IP-based facilities according to the investment plans for each operator. As the PSTN's IP conversion exceeds a certain level and VoIP; (an alternative service); is activated, some countries; such as Europe; are considering terminating the PSTN service, centering on operators. In this paper, trends in the procedure, timing, and major issues related to the termination of an overseas PSTN are examined. The domestic status is also examined.

• 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.

• Journal of Internet Computing and Services
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• v.4 no.5
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• pp.77-86
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• 2003

This paper introduces the extension of SIP protocol for Multiparty Conference and the implementation of IP-based Multi-Conference. SIP protocol is a signaling protocol for initiating, modifying and terminating interactive sessions (voice, video, text, application). Multiparty conference system is implemented by RTP protocol for real time transmission and by H.323 for call setup. As H.323 fits into PSTN, it has some problems (call setup delay, hard implementation) for IP applications. IETF developed SIP protocol used for signaling in IP networks. However this SIP protocol doesn't explain signal protocol about multimedia conference, which is different from the existing H.323. So this paper describes the extension of SIP according to SIP Specification and case study for the multimedia conference.