• 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.

• 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.

• Journal of Internet Computing and Services
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• v.10 no.6
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• pp.27-36
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• 2009

PSTN (public switched telephone network) provides voice communication service, whereas IP network provides data oriented service, and we can use IP network for multimedia transport service (e.g. voice over IP service) with economic price. In this paper, we propose RoIP (robot on IP) service scenario, signaling call flow, and implementation to provide home automation and monitoring service for remote site users. In our scheme, we used a extended SIP (session initiation protocol) for signaling protocol between remote site users and home robots. For our bearer transport control, we implemented H.263 video codec over RTP (real-time transport protocol) and additionally DTMF (dual tone multi-frequency) transport for robot actuator control. We implemented our scheme on home robots and experimented with KTF operator network, and it shows good communication quality (average MOS = 9.15) and flexible robot controls.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.1
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• pp.75-82
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• 2003

We can foresee the rapidly growing of users by development of VoIP service what can transmit a audio traffic with low cost among a lots application using internet. VoIP needs a standardized protocol that is able to do signaling for offering high quality of services such as mobility, universal number, multiparty conference, voice mail, automatic call distribution. At the present time, a base composition elements of SIP(Session Intiation Protocol) are developing for offering VoIP based SIP in the inside and outside of the country, because SIP of IETF which has a strength from 'fast connection', 'parsing' & 'easy to compile' points of view. This paper suggests a type of Hybrid SIP Server for providing some services as 'a reducing load of SIP server that process a request method from users', 'efficiency of managing networks', 'offering services to many users'.

• 한국정보통신설비학회:학술대회논문집
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• 2003.08a
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• pp.223-225
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• 2003

Session Initiation Protocol(이하 SIP)은 H.323 network에서 제기된 문제점을 해결하고 다양한 multimedia service를 제공하기위해 IETF에서 정의된 protocol로서, 3GPP의 IP Multimedia Subsystem(이하 IMS)의 기반 protocol로 채택되었다. IMS는 multimedia service를 제공하기 위한 infrastructure로서 CSCF, HSS, MRF, MGCF, IM-GW등 기능적으로 분리된 여러 entity로 구성되며, 향후 추가될 다양한 부가 service에 유연하게 대처할 수 있는 Application Server(이하 AS) mechanism을 포함하고 있다. 현재 AS mechanism을 이용하여 3GPP규격 상에 정의된 대표적 서비스는 presence service이며, presence service network은 presence server, presence agent, presence proxy, watcher등으로 구성된다. 마지막으로 상기에 언급한 규격을 바탕으로 실제 구축된 Instant messaging/presence service network을 제시한다. (본문 참조)

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1B
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• pp.101-108
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• 2004

SIP(Session Initiation Protocol) is a suitable protocol for supporting Internet telephony services and currently market requirements fur developing supplementary telephony services such as unconditional call forwarding, call forwarding on busy or no answer, call filtering services have recently grown. CPL(Call Processing Language) is a standard technology that can be used to describe and control internet telephony services. In this paper, we describe the CPL system for supplementary Internet telephony services using SIP as an application level call signaling protocol. Those supplementary services are composed of CPL client which is a SIP UA, SIP Proxy server, Registrar and CPL server In this paper, we describe the design and implementation of the CPL server system in detail which is developed in Linux 7.2 using C and C++ programming languages.

• Journal of the Korea Society of Computer and Information
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• v.13 no.7
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• pp.199-205
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• 2008

The centralized conference model has advantage of conference management and control. however it's scalability has been limited as performance degrades largely with increasing number of conference users. So new distributed conference models which improve scalability of centralized conference model have been suggested recently. In the distributed conference model. as conference users exceed a predefined maximum number, a new conference server is added to the conference dynamically. In this paper, We have proposed a new method which increases efficiency of conference event package processing that primary conference server should charge in the distributed conference environment. The primary conference server exchanges informations with each secondary conference servers and conference users by using conference event package. And from the conference information database it selects SIP(Session Initiation Protocol) UA(User Agent) which will share notification to the conference users, and transfers lists to each conference servers. The conference servers make the selected UAs share processing of conference event package, so loads of SIP signal processing decrease, and improve scalability of distributed conference model. The performance of our proposed model is evaluated by experiments.

• Journal of the Korea Society of Computer and Information
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• v.14 no.5
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• pp.85-91
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• 2009

The centralized conference architecture has a restriction in scalability due to the performance reduction as the number of conference participants increases. To solve this problem several distributed conference architectures have been studied recently. In these architectures new conference servers are added dynamically to the conference environment. In this paper, We have proposed a new conference information data model which can be used in these distributed conference architectures. In our newly proposed conference information data model. several components has been added for exchanging conference information between primary conference server and multiple secondary conference servers. We also proposed a procedure of conference information exchange between these conference servers. And the management of conference informations and SIP(Session Initiation Protocol) notifications to the total conference participants can be processed distributedly with these conference servers, therefore the load to the primary conference server can be decreased by using this method. The performance of our proposed model has been evaluated by experiments.

• The Magazine of the IEIE
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• v.28 no.6
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• pp.80-87
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• 2001

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.12
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• pp.2533-2540
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• 2009

Performance of processing protocols is one most important goal in a network or systems constituting the network. Measurement of performance, thus, is an essential element to not only establish the network but also develop systems. A projection in vector mathematics is the transformation of points and lines in one plane onto another plane by connecting corresponding points on the two planes with parallel lines. This is a method, as an application of vector mathematics, which is widely used in engineering as well to consider elements dedicated to the measurement object. This paper proposes a sound methodology for measuring the performance of the SIP protocol, which is based on the projection. The SIP protocol is a typical standard protocol for call-processing in the internet telephony of Next Generation Convergence Network. Owing to use the projected protocol which excludes unnecessary operation paths, the methodology can be effective in implementation and resource utilization. This paper also presents a process based on the proposed methodology to measure the performance of a SIP proxy server.