• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.3B
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• pp.195-207
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• 2011

In this paper, we design and implement a visual IMS protocol analyzer, VIPA(Visual IMS Protocol Analyzer), providing a graphical view of session connection information between user equipments on IMS network. The VIPA captures SIP packets, which are used to setup sessions on IMS networks, and shows graphical view of session information to provide easy way to analyze IMS protocols. The existing protocol analyzer has limitation to show only terminal side protocol analysis information. Whereas the VIPA can provide not only terminal side but also server side analysis result so that the connection status between all the session participating user agents can be monitored.

• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.95-104
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• 2002

The interworking for voice service between next generation mobile network and traditional network can be deployed better flexible and expansible network conditions with providing efficiency and economy of network at the same time. So it is essential to develop the interworking strategies together with a evolved network and traditional network. This paper describes a design and verification of internetworking protocol for CC of next generation mobile network and SIP of IP network for applying interworking technology to next generation mobile network, which can harmoniously expropraise voice service from traditional network.

• Journal of Digital Contents Society
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• v.16 no.1
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• pp.173-177
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• 2015

This paper proposed an access control for computer-based integrated multimedia running on shepherd and SIP(Session Initiation Protocol). But, conventional framework for access control SIP environment has not yet fully progressed a shepherd and an access control for computer-based integrated multimedia running on SIP(Session Initiation Protocol). Session management include function of session creation, session end, late comer process, and access control. Therefore, this paper described access control based on a shepherd and SIP environment to maintain good session condition.

• Journal of KIISE:Information Networking
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• v.34 no.1
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• pp.73-80
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• 2007

This paper presents an approach to support dynamic QoS(Quality of Service) requirements of largely emerging SIP(Session Initiation Protocol) multimedia applications in broadband access networks. The topology of QoS-enabled broadband access networks and its operational model to support SIP QoS are firstly suggested. And then the procedures to bind QoS-enabled SIP signaling into an IP QoS mechanism are presented. In this paper, DiffServ-based IP QoS architecture is deployed due to the complexity problem of the other IntServ architecture, and COPS(Common Open Policy Service) and COPS-PR protocol based signaling mechanisms are used to support dynamic DiffServ QoS, correspondent with dynamic SIP QoS. The broadband access network is assumed to support rapidly expanding Metro Ethernet 802.1 D/Q QoS, and how to translate SIP QoS parameters into IP DiffServ classes and DiffServ classes into 802.1 D/Q QoS parameters is also presented in this paper.

• Journal of the Korea Society for Simulation
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• v.19 no.4
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• pp.43-50
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• 2010

Recently, the Internet telephony has been used rather than the traditional telephony by many Internet users, with low cost. Session Initiation Protocol(SIP) is the standard of application layer protocol for establishment and disconnection of the session for Internet telephony. SIP mainly runs over the UDP for transport. So in case of the loss of the INVITE request message, the message is retransmitted by an appropriate timer for reliable transmission of the UDP message. Though the retransmission is useful for ensuring the reliability of SIP messages sent by the users, it may cause the overload traffic in the SIP proxy server. The overload in SIP proxy servers results in the loss of many input messages. This paper presents a non-linear overload control algorithm to resolve the overload condition of the server. we simulate our proposed algorithm using the network simulator ns-2. The simulation results show that the throughput of the server with the proposed algorithm have been improved about 12% compared to the existing linear control algorithm.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.4
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• pp.39-44
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• 2014

IMS(IP Multi-Media Subsystem) base on the IP service platform which can offer multimedia as the voice, audio, video, and data is service platform. In 3G mobile communication in the early day, IMS had a suggestion for supporting to multimedia service in the 3GPP. But now It is broadly substituting in the IPTV, wire phone company and it is substituted in internet platform base on the soft-switch in currently. Especially nowadays, 4G LTE in a mobile communication company is rapidly growing in market. Therefore, in this study, we had designed to the main prosser that can admit to 1,000 user over and SIP gateway which can link the IMS 코어 that can link SIP Device which adopt the standard protocol on the SIP.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.12
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• pp.2301-2306
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• 2008

SIP(Session Initiation Protocol) has become an universal standard for multimedia communications for both wired and wireless networks since it has been adopted as a standard protocol for IMS platform in 3GPP standardization organization at November 2000. In this paper, we design and implement a SIP-based softphone client program which provides telephony service between internet users and a call center equipped with VoIP gateway. A softphone client based on PC-to-phone connection should guarantee to provide interoperability with various VoIP gateways and higher portability to be able to operate on different PC environments. The softphone client program in this paper has been developed with SIP 2.0 standard protocol to support interoperability and with JAIN SIP and JMF package to achieve higher portability.

• Proceedings of the Korean Association of Geographic Inforamtion Studies Conference
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• 2008.10a
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• pp.65-72
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• 2008

Mobile Collaboration is an enabling technology that makes users share information between mobile devices and various Mobile P2P platforms have been designed and implemented for it. There are, however, few research papers on application of SIP protocol to spatial data sharing on mobile devices. In this paper, SIP based real time sharing framework is proposed to compose a mobile P2P platform on which spatial data can be trans(erred. A new protocol based on WKT and WKB is defined to send and receive spatial objects with SIP MESSAGE method. Base maps such as digital maps and parcel maps can be provided by a map server that is integrated with SIP server after a new SIP session established and client agents are registered. The framework proposed based onSIP enables users to transfer spatial data such as maps and satellite images directly between mobile devices during VoIP based voice call and therefore, mobile applications can be applied in various domains such asforest management and national defense.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.283-285
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• 2002

본 논문에서는 현재 VOIP(Voice Over IP)기술고서 ITU-T의 H.323에 대응되는 텍스트 기반 프로토콜인 SIP(Session Initiation Protocol)를 구현한 오픈 소스인 VOCAL(Vovida Open Communication Application Library) 시스템의 구조를 분석하고 SIP의 시스템을 구성하는 여러 모듈들을 테스트를 통하여 검증하였다. VOCAL의 사용자 에이전트에 RTP(Real-time Transport Protocol ) 포맷의 데이터를 전송/수신할 수 있는 모듈을 구현하고 검증하여, SIP 시스템에서 사용할 수 있는 사용자 에이전트의 활용에 대한 방안을 제시하였다.

• Proceedings of the Korea Multimedia Society Conference
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• 2002.05c
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• pp.455-458
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• 2002

단말간 호설정 기능을 제공하는 SIP (Session Initiation Protocol)는 VoIP 표준 기술로 인식되고 있다. 그러나 SIP는 단대단 프로토콜이므로 새로운 응용으로 다자간 회의 시스템에 적용하기 위해서는 새로운 메소드 추가 및 새로운 헤더 추가 등이 요구된다. 특히, 멀티미디어 회의 시스템을 위해서는 오디오는 물론 비디오 관련 정보도 협상되어야 한다. 본 논문에서는 현재 IETF에서 진행되고 있는 SIP 기반 다자간 멀티미디어 회의 시스템을 분석한다.