• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.682-685
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• 2003

SIP(Session Initiation Protocol) 프로토콜은 구조가 간단하고 다양한 애플리케이션을 개발할 수 있는 등의 장점을 가지고 있기 때문에 차세대 네트워크에서 호 설정을 위한 사실상의 표준으로 자리 잡아가고 있다. SIP는 계층적 구조를 갖는 프로토콜로서, SIP의 일부 계층은 대부분의 SIP 응용 애플리케이션에서 공통적으로 사용될 수 있다. 본 논문에서는 SIP 애플리케이션 개발에 효과적으로 사용될 수 있는 SIP 공통모듈에 대한 간략한 설명과 이를 이용한 사용자 에이전트의 설계 및 구현에 대하여 설명한다.

• 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.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.139-141
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• 2003

VoIP(Voice over IP) 기술들 중에서 IETF(Internet Engineering Task Force)에서 제안한 텍스트 기반의 SIP(Session Initiation Protocol) 프로토콜은 다양한 애플리케이션을 개발할 수 있는 등의 장점을 가지고 있기 때문에 차세대 네트워크에서 호 설정을 위한 사실상의 표준으로 자리 잡아가고 있다. SIP는 계층적 구조를 갖는 프로토콜로서, SIP의 일부 계층은 대부분의 SIP 응용 애플리케이션에서 공통적으로 사용될 수 있다. 본 논문에서는 SIP 애플리케이션 개발에 효과적으로 사용될 수 있도록 설계 및 구현된 SIP 공통모듈에 대하여 설명한다.

• Journal of Internet Computing and Services
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• v.13 no.3
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• pp.31-47
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• 2012

The mobile Virtual Private Network (MVPN) of Internet Engineering Task Force (IETF) is not designed to support NEwork MObility (NEMO) and is not suitable for real-time applications. Therefore, an architecture and protocol which supports VPN in NEMO are needed. In this paper, we proposed the cost-reduced secure mobility management scheme (CR-SeMMS) which is designed for real-time applications in conjunction with VPN and also which is based on the session initiation protocol (SIP). Our scheme is to support MVPN in NEMO, so that the session is well maintained while the entire network is moved. Further, in order to reduce the authentication delay time which considers as a delaying factor in hands-off operations, the signaling time which occurs to maintain the session is shortened through proposing the hands-off scheme adopting an authentication method based on HMAC based One Time Password (HOTP). Finally, our simulation results show the improvement of the average hands-off performance time between our proposed scheme and the existing schemes.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.11b
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• pp.745-748
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• 2003

본 논문에서는 SIP(Session Initiation Protocol) 호의 처리 모듈을 가지고 있지 않은 웹 상의 클라이언트들에게 음성서비스 및 멀티미디어 서비스를 제공할 수 있는 SIP 호 처리 서버를 설계 및 구현 하였다. 서버상에서 SIP 호 설정을 위해 사용되는 메시지를 VoiceXML의 태그형태로 변환하는 기능을 제공함으로써 비 SIP 사용자들에게도 SIP 서비스를 할 수 있도록 하였다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.5
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• pp.1144-1150
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• 2015

Presence information provides various informations about users such as on-line status, current location, network connection method and connection address, and there are two kinds of presence information, SIP(Session Initiation Protocol) based presence information and XMPP(Extensible Massaging and Presence Protocol) based presence information. In this paper, a multiple server architecture that can handle these two kinds of presence information has been proposed. In this architecture, severs are added dynamically according to number of users to provide system scalability, and load of each server can be effectively controlled. In this system, a new XMPP stanza architecture and presence information data format are designed for load control. Furthermore message exchanging procedures between servers and users for dynamic server control has been also suggested. The performance of the proposed system has been analysed by simulation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39C no.1
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• pp.90-96
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• 2014

IEEE802.15.4 that is a standard for sensor networks is mainly used the wireless personal area networks such as ZigBee networks and it features low-power, low-speed data communication. However, recently research for interworking sensor network based voice communication and Session Initiation Protocol (SIP) for long-range, multi-user support has been actively conducted. In this paper, we designed a integrated base station based existing systems for interworking sensor networks based voice communication and SIP. We measured number of packet and delay according to increase the number of users to evaluate the performance of designed Base Station.

• The KIPS Transactions:PartC
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• v.15C no.2
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• pp.87-92
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• 2008

This paper proposes a session control model for the P2P(Peer to Peer) IPTV(Internet Protocol Television) services and presents the IPTV session control procedures based on the proposed model. Since, while public IPTV traffic is usually processed via a separate network, P2P IPTV traffic is processed together with the conventional Internet access traffic, the P2P IPTV control mechanism needs to provide multi-stream processing for the constituent TPS(Triple Play Service) traffic and corresponding QoS(Quality of Service) control functions. Besides, P2P IPTV session control mechanism should provide appropriate multicast control functions in order to support effective transmission of video traffic generated by personal IPTV broadcasters. The P2P IPTV session control model proposed in this paper is designed to be based on the standard SIP(Session Initiation Protocol), IGMP(Internet Group Management Protocol), and COPS(Common Open Policy Service) protocol so that it can contribute to the easy and prompt deployment of inter-operable P2P IPTV platform.

• The KIPS Transactions:PartC
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• v.13C no.4 s.107
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• pp.461-470
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• 2006

Recently mobile Stream Control Transmission Protocol (mSCTP) has been proposed as a transport layer approach for supporting mobility. When a mobile terminal (MT) is not located in the home network. a terminal that wishes to communicate with the MT is not able to establish mSCTP association to the MT, since mSCTP does not include the location management mechanism. In order to solve this problem. an interworking approach using the Session Initiation Protocol (SIP) INVITE method has been proposed. However, this approach has shown subsequent delay in acquiring the current location information of the MT when initiating mSCTP association establishment. In this paper, we propose new SIP methods and an approach that minimizes the address acquisition delay (AAD) by utilizing those SIP methods. Mathematical analysis and simulation results show that the proposed approach is more efficient than the previous approach in terms of AAD in all kinds of SIP environments.

• The Journal of the Korea Contents Association
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• v.8 no.5
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• pp.1-8
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• 2008

The centralized conference architecture, one of the conference architectures in SIP(Session Initiation Protocol) environment, is widely used as it has the advantage of conference management and control. However it has been limited in scalability. Therefore we have proposed an efficient distributed conference architecture to improve scalability of centralized conference model. In our architecture, if the number of conference participants exceeds the predefined maximum number, a new conference server is added to the conference dynamically. In this case, the focus of existing server acts as primary focus and the focus of added server acts as secondary focus, and dynamic reallocation of participants between servers is done to equally divide the loads. This process is repeated as the number of conference participants increases. For this behavior, we have proposed procedure of adding the conference server, SIP call signal exchange, signaling procedure for RTP(Real Time Transport Protocol) sessions between conference servers, and procedure of conference event package between conference servers. The performance of our proposed model is evaluated by experiments.