• Proceedings of the Korean Information Science Society Conference
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• 2004.10c
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• pp.331-333
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• 2004

본 논문에서는 SIP(Session initiation Protocol) 기반의 VoIP(Voice over Internet Protocol) 시스템에 효율적인 call-control을 위해 필요한 헤더와 파라미터를 추가한 Extension SIP를 제안하였다. 또한 이 제시된 Extension SIP에 따르는 SIP Proxy Server와 User Agent(User Agent Client, User Agent Server)를 리눅스 시스템에서 C언어를 통해 구현하였고, 이 구현된 Extension SIP 시스템을 통해 기존의 SIP 시스템과 cail-control을 위한 packet traffic을 비교.분석 함으로써 제안한 Extension SIP의 효율성 을 증명하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.176-179
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• 2003

The use of Multiparty Conference service based on SIP for VoIP provides is gradually magnified, and the work for continuous development and standardization on SIP is in the process of advancing. But, currently it is impossible for SIP to support identity discovery and distribution of each participant for multiparty conference. In this paper, we propose a SIP extension for guaranteeing security on the multiparty conference using SIP by adding new method and reconstructing header informations. With this, it is also possible to identify discovery and to distribute each participant using SIP extension when a call is established for closed multiparty conference.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.2
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• pp.331-337
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• 2004

Multiparty conference service based on SIP supported by VoIP network is gradually increased in use and the continuous development and standardization works on SIP are in the process of advancing. But SIP used in currently does not support identity discovery and distribution of each participant for multipath conference. In this paper, a SIP extension method for guaranteeing security from the multiparty conference based on SIP is proposed. We design a new SIP header and method for discovering and distributing a participant's identity in closed multiparty conference when the call initiation is established. And it can ensure that each participant is notified before a new participant joins.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.448-450
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• 2002

본 논문은 한국전산원 차세대 인터넷 응용서비스 2차 과제로 선정된 "개인 이동성을 지닌 지능형 웹 컨퍼런스 개발"을 수행함에 따른 결과를 정리했다. 일반적으로 인터넷 전화 프로토콜의 신호 프로토콜로 H.323과 SIP를 사용하고 있다. 최근 들어 H.323의 몇몇 낙제점 때문에 인터넷에 적합한 SIP로 대치되고 있는 상황이다. 그러나 SIP는 기존 H.323과 달리 멀티미디어 컨퍼런스에 관한 신호 프로토콜이 명시 되어 있지 않다. 본 논문에서는 이런 SIP의 문제점을 해결하기 위하여 멀티미디어 컨퍼런스를 위한 SIP 확장에 대해 기술한다.

• 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.

• 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.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10c
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• pp.580-582
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• 2002

최근 인터넷 관련 기술의 급속한 발전으로 데이터, 음성, 화상 등의 다양한 멀티미디어 서비스가 통합된 개방형 네트워크로 진화되어지고 있다. 모든 미디어가 인터넷으로 수렴되는 NGN(Next Generation Network)으로 발전할 전망이다. 개방형 네트워크는 다양한 유무선 통합망의 융합화에 따른 통신망간의 간섭이 증가하고 네트워크 접속점 중심의 통신망간 접속구조가 확대되어 지금까지의 시스템 보안 위주의 단순한 보안 기술을 적용하기가 어려웠다. 따라서 네트워크 노드간을 효율적으로 보호하는 네트워크 중심의 보안 기술이 필요한 시점이다. 이에 본 논문은 IETF에서 제안한 텍스트 기반 응용 계층의 접속제어 프로토콜로 실시간 미디어 통신을 위한 차세대 인터넷 프로토콜로 주목받고 있는 SIP(Session Initiation Protocol) 시스템을 분석하고, SIP 기반의 보안 메커니즘인 IPSec에서의 확장 보안 메커니즘을 설계하였다.

• The Journal of the Korea Contents Association
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• v.7 no.3
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• pp.34-41
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• 2007

The presence service which provides user's presence information by subscription and notification is one of SIP(session initiation protocol) extension services, and it is used importantly in VoIP(Voice over IP) and Instant Messaging service. In this paper, we propose a new method in which users can combine and control presence service and call processing services in various ways by extending call processing language, and only changed parts of the presence information are published instead of full presence information document. Each user registers full presence information document with his own call processing script during the first publication to a presence server. The presence server executes these call processing scripts, so it can provide various services with combination of presence service and call processing services during the presence subscriptions and notifications. Afterwards, users publish only changed parts of the presence information and the presence server notify only these changed parts to watchers. Therefore the efficiency of the overall system can be improved. The performance of our proposed model is evaluated by experiments.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.1
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• pp.1-7
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• 2008

Today, Korean medical community faces rapid changes in medical environment due to opening of medical market, more emphasis on making profit and introduction of private medical insurance. Therefore, it is apparent that around, major university hospitals, efforts are being made to adapt to such changes by establishing mid to long range strategic plans. We want to keep pace with changing times and diverse demand of patients by introducing state of the art system, utilizing Ubiquitous technologies for improvements. In doing so, we want to distinguish our hospital services from others. However, Hospital Information System that integrates ubiquitous technologies are introduced in limited basis due to problems like standardization and limits on medical use etc. Particularly, problems like absence of tag design suitable for medical environment, compatibility and extension issue with RFID system need to be addressed on application of RFID technologies. In order to solve such problems, this paper implemented RFID tag system, RFID-SIP UA program, location tracking program. This paper provides a comprehensive basic review of RFID and SIP model suggests the evolution direction of further advanced RFID application service. The design of RFID-SIP application offers advance RFID system with drawback and reduction search time of medical object. so there is a improvement of hospital information system in ubiquitous environment.

• Journal of Communications and Networks
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• v.9 no.4
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• pp.511-523
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• 2007

In broadband convergence network(BcN), heterogeneous broadband wired & wireless subnetworks and various terminal equipments will be interconnected. In order to provide end-to-end realtime multimedia services on such heterogeneous networking environment, as a result, two major problems should be resolved: i) Multimedia session establishment & negotiation that adjusts the differences in the capability of multimedia data processing at the end terminal nodes, ii) quality of service(QoS)-guaranteed connection establishment or resource reservation with connection admission control(CAC) in each heterogeneous subnetworks along the path. The session layer signaling(e.g., SIP/SDP) should be extended for QoS negotiation, and must be tightly cooperating with network layer signaling or resource reservation with CAC function. In this paper we propose a session and connection management architecture for the QoS-guaranteed realtime multimedia service provisioning on BcN, with Q-SIP/SDP, resource reservation protocol with traffic engineering extension, and CAC functions. The detailed interaction scenario and related algorithms for QoS-guaranteed realtime multimedia session, resource reservation and connection establishment are explained and analyzed. From the experimental implementation of the proposed scheme on a small scale BcN testbed, we verified that the proposed architecture is applicable for the realtime multimedia service provisioning. We analyze the network performance and QoS parameters in detail.