• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.12
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• pp.292-305
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• 2013

EPCglobal has proposed the EPC Sensor Network to efficiently integrate and interoperate the RFID and WSN technologies based on the international standard EPCglobal Network. However, it can not deal with the sensor data without EPS. Also, it does not provide enough management function for the sensor data only. This study proposes USN management system that can deal with these weaknesses of EPC Sensor Network. Not only that, it is suitable for the medical environment properly. In the medical environment, it is important for the medical information to be integrated and interconnected via the standard for medical data. This proposed USN management system is based on SIP-HL 7 and can send RFID and WSN data based on SIP(Session Initiation Protocol). It can be interconnected with the existing hospital information system and exchange the information. Also it can design and materialize SIP Interface based on HL7 that allows the medical information to be exchanged and sent.

• Journal of the Korea Society of Computer and Information
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• v.18 no.4
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• pp.79-85
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• 2013

In SIP(Session Initiation Protocol) environment, the presence server should process SIP SUBSCRIBE request messages including multiple presence resources addresses from users, and also precess massive notification messages from the subscribed presence resources. The load of the presence server increases as number of users increase, and it limits system extendability. Therefore a new multiple presence servers architecture has been suggested in this research. In this architecture presence servers can be added dynamically and each server's load can be controlled effectively as number of users increase. Each presence server can monitor current load status of entire presence system by using presence event notification package which newly has been suggested in this paper. When a particular presence server's load increases over predefined limit, the presence service processing is distributed by selecting a server which has the smallest load, or by generating a new server dynamically. In this system the overall load of the entire system can be controlled optimally and extendability of the system can be increased. For this purpose a new presence event notification package and presence information data format have been suggested. The performance of the proposed system has been evaluated by experiments. They shows 44.3% increase in SUBSCRIBE message processing time, and 43.1% increase in Notification message processing time.

• Journal of the Korea Society of Computer and Information
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• v.16 no.9
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• pp.123-129
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• 2011

In this paper, We have proposed a new distributed conference architecture which can reduce largely SIP(Session Initiation Protocol) messages exchanged between servers and participants in distributed conference environment by using resource lists which consist of conference participants URIs. This architecture can also reduce processing time and loads of conference servers. In our proposed method, the number of SIP messages required for participating in the conference can be largely reduced, and the number of SIP messages required for subscription of conference event package and maintaining sessions can also be largely reduced by using resource lists. Therefore expandibility of the conference system increases and network traffic can be reduced by using our conference architecture. And We also have proposed in this paper a new conference information data format with some added elements that are necessary for processing resource lists. Furthermore the procedure for dynamically adding conference servers to the conference environment has been suggested, and the exchanging process of SIP messages and conference information data between conference servers and conference participants has been also suggested. The performance of our proposed system has been evaluated by experiments.

• KIPS Transactions on Computer and Communication Systems
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• v.3 no.4
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• pp.115-124
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• 2014

Since it costs a lot to build a home network system of new service provider, the entry barrier is quite high. If it would be possible to build a home network system with high performance at a low building cost, it would be quite attractive. If a home network would be built in P2P-SIP(Peer to peer - Session Initiation Protocol) structure, it is possible to decrease server maintenance cost and keep high usability by deconcentrating the traffic concentrated on the server. It is also possible to efficiently manage the location information of terminal, if quorum system based on fuzzy logic would be applied. In this paper we propose the P2P-SIP structure applied of quorum system based on fuzzy logic by which a home network system can be built at low cost. It was possible to know by comparing the structure with existing one that the structure has very good performance.

• Proceedings of the Korean Information Science Society Conference
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• 2003.04d
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• pp.542-544
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• 2003

인터넷이 생활의 일부분으로 변모한 지금 사용자들은 음성 및 영상 또는 텍스트 메시지와 같은 다양한 실시간 멀티미디어 세션의 전송을 위한 다양한 프로토콜을 필요로 하고 있다. 이에 SIP(Session Initiation Protocol)는 User Agent라는 인터넷 종단점을 발견하고 상호간에 세션의 특징을 그대로 사용할 수 있게 함으로써 다양한 사용자들의 욕구에 부응할 수 있는 차세대 통신 프로토콜로 각광받고 있다. 이에 본 논문에서는 SIP User Agent를 구현하는 방안과 설계 및 구현하는 과정을 보이고자 한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.11b
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• pp.1419-1422
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• 2002

SIP(Session Initiation Protocol)는 기존의 VoIP(Voice over IP)를 위한 시스템에서 뿐만 아니라 호설정의 처리와 제어가 필요한 여러 인터넷 응용분야에 적용이 가능한 확장성이 뛰어난 프로토콜이다. 또한 HTTP(Hyper Text Transfer Protocol)와 유사한 텍스트 기반의 응용계층의 프로토콜로서 호처리시 절차가 비교적 간결하다. 본 논문에서는 이러한 SIP의 장점을 이용하여 화상회의를 위한 시스템을 구현하였다. 이때. 인증 기능을 부여하여 화상회의시 세션 연결을 위한 호설정을 이루는 과정에서 상대방을 인증할 수 있도록 하였고 인증방법으로는 메시지 인증기능과 함께 리플레이 공격(replay attack)의 방지기능을 가진 SIP 다이제스트 인증(Digest Authentication) 방법을 사용하였다. 본 화상회의 시스템에 사용된 SIP 사양은 최근에 발표된 RFC 3261을 기준으로 하였다.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.11b
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• pp.1205-1208
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• 2002

현재 VoIP(Voice over IP) 기술의 발전을 주도하고 있는 SIP(Session Initiation Protocol)은 IETF에서 표준화된 프로토콜로서 인터넷 컨퍼런스와 인터넷 텔레포니 등 멀티미디어 응용을 위한 기술이다. 이 프로토콜은 텍스트 기반의 시그널링을 통하여 넓은 확장성과 다양한 응용성을 제공하면서 기존의 H.323 을 대체하는 기술로서 주목을 받게 되었다. 그러나 사실상 현재의 인터넷 기술을 이용하여 기존의 음성 전화서비스와 동일한 품질의 서비스를 제공하기에는 아직도 해결해야 할 기술들이 많은 실정이다. 특히, 서비스 품질(Quality of Service)을 보장하는 문제는 VoIP 기술의 성장을 위해서 필수적이다. 이에 본 논문은 QoS 제공 망에서 SIP UA 가 QoS 가 제공되는 콜(call)을 설정하기 위하여 확장되어져야 하는 모듈 및 메소드와 변화되는 시그널링 절차등에 대하여 설명하고, 이를 실제 VoIP 응용에 적용하기 위한 설계 및 구현 작업에 대하여 기술한다.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.292-294
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• 2001

VoIP(Voice over IP) system은 현재 크게 2가지 형태로 진행되어가고 있다. 첫 번째는 H.323을 이용한 방법이고, 두 번째는 SIP(Session Initiation Protocol)를 이용한 방법이다. H.323은 실제 데이터를 전송하기전 호처리에 많은 signaling이 이루어지는 관계로 SIP보다 많은 RTT(Round Trip Time)를 소모하게 된다. 따라서 매우 복잡하고, LAN환경을 바탕으로 만들어서 확장성면에서도 여러 문제점을 가지고 있다. 그래서 본 논문은 호처리는 SIP를 이용하고, 실제 음성전송은 RTP(Real-Time Transport Protocol)와 RTCP(RTP Control Protocol)를 이용하는 시스템 구현을 제시한다. RTP는 실시간 특성을 가지는 데이터에 대해서 종단간 전송 서비스를 제공해주는 프로토콜로, 어떠한 인코딩에도 적합한 프레임워크를 제공한다. 그런데, RTP는 완전한 하나의 프로토콜이 되기 위해서는 RTP와 페이로드 포맷이 함께 제공되어야 하므로, 구현시스템은 음성신호를 PCM(Pulse Code Modulation), ADPCM(Adaptive Differential PCM)등의 여러 압축기술을 이용하여 파일을 생성하여 실시간으로 RTP와 RTCP를 이용하여 전송하는 방법을 제시한다.

• 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.

• 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.