• Proceedings of the Korea Multimedia Society Conference
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• 2001.11a
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• pp.41-45
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• 2001

본 논문에서는 인터넷 망을 통해 음성 데이터를 전달한 수 있는 VoIP 기술과 인도어(indoor) 환경에서 사용자에게 이동성을 제공하는 블루투스 기술을 연동하여 새로운 VoIP wireless phone을 설계한다. 또한 본 논문에서 제안한 VoIP wireless phone의 호 접속제어를 위하여 새로운 프로토콜과 프로토콜내에서 사용되는 메시지를 설계한다.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.17 no.2
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• pp.61-72
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• 2007

With the increasing growth of Internet and wireless IP networks, Multimedia systems need to be envisaged as information resources where users can access anywhere and anytime. However, efficient services in these multimedia systems are open and challenging research problem due to user mobility, limited resources in wireless devices and expensive radio bandwidth. To implement multimedia services over heterogeneous network, the IP header compression scheme can be used for saving bandwidth. In this paper, we present an efficient solution for header compression, which is modified form of ECRTP. It shows an architectural framework adopting modified ECRTP when IP tunneling network using GRE over IPSec is implemented. We have conducted simulations in order to analyze the effects of different header compression techniques while delivering real-time services to the wireless access network through secured IP Network. The impacts on performance have been investigated through a series of experiments.

• 한국정보통신설비학회:학술대회논문집
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• 2009.08a
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• pp.91-96
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• 2009

In Public Switched Telephone Networks(PSTN) number portability is implemented by utilizing Intelligent Network(IN) functions for number mapping. And voice over IP(VoIP) and IP Multimedia Subsystem(IMS) networks can deploy number portability by using E.164 Number Mapping(ENUM). This paper discuss the possibility of using E.164 Number Mapping(ENUM) for number portability in voice over IP/IP Multimedia Subsystem and Public Switched Telephone Networks, eliminating the need for Number Portability Database(NPDB) for number portability routing data in Public Switched Telephone Networks.

• Proceedings of the Korea Multimedia Society Conference
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• 2000.04a
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• pp.397-400
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• 2000

인터넷에서 음성 서비스를 제공하는 IP 텔레포니 또는 VOIP(Voice over IP)기술은 대부분 ITU-T H.323을 기반으로 제공되고 있다. 그러나 H.323은 그 구조가 복잡하기 때문에 이해하는데 상당한 노력과 오랜 개발 기간이 요구된다. 특히 표준을 따라 개발한 제품간에 상호연동을 위해서 상당한 노력이 필요하다. LFTF는 이러한 문제를 극복하고 인터넷 환경에서 잘 동작할 수 있는 IP 텔레포니용 프로토콜을 표준화하고 있으며 , 본 논문에서는 이들 프로토콜을 분석한다.

• Proceedings of the Korea Multimedia Society Conference
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• 2000.11a
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• pp.299-302
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• 2000

인터넷에서 음성 서비스를 제공하는 인터넷 텔레포니 또는 VoIP(Voice over IP) 기술은 대부분 ITU-T H.323을 기반으로 제공되고 있다. 그러나 H.323은 그 구조가 복잡하기 때문에 이해하는데 상당한 노력과 오랜 개발 기간이 요구된다. IETF는 이러한 문제를 극복하고 인터넷 환경에서 잘 동작할 수 있는 IP 텔레포니용 프로토콜로서 Session Initiation Protocol (SIP)을 표준화하였다. 본 논문에서는 VoIP 서비스를 위한 SIP의 구현 사항을 기술한다

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.11B
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• pp.693-699
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• 2007

In this Paper we present analysis of efficiency for various encapsulation methods to transport IP datagram over terrestrial digital multimedia broadcasting (T-DMB). The multimedia broadcasting system transmitted in one-way needs the data reliability for efficient transmission. And then It is able to select the independent data service using If encapsulation over T-DMB. We examine the compatibility in adopting T-DMB and the overhead of transmission occurring encapsulation. it also needs the full analysis of data framing because the cost of return channel is relatively higher. We do analysis based on theoretical calculation and propose an efficient, reliable and adaptable method in T-DMB.

• Journal of Korea Multimedia Society
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• v.10 no.3
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• pp.335-347
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• 2007

VoIP (Voice over IP) is a technology to transport video and voice traffic over IP networks such as Internet. Today, the VoIP technology is viewed as the right choice for providing voice, video, and data communication among various terminals over the next generation network. This paper discusses a multipoint VoIP implementation with end-point mixing model which can support multipoint conference without a conference bridge. The multipoint VoIP is implemented with SIP (Session Initiation Protocol), and supports STUN (Simple Traversal of UDP Through NATs) since it works in an asymmetric NAT (Network Address Translator) environment. The characteristics of this paper are as follows. It is possible that all terminals in the hierarchical conference don't receive the duplicated media information because we use the end-point mixing model with the new media processing module. And, the paper solves the problem that the hierarchical conference session should be separated into several sessions when a mixing terminal terminates the hierarchical conference session.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.16 no.7
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• pp.4878-4883
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• 2015

The IP Multimedia Subsystem(IMS) is a framework that provides access to the content of Internet and Telecom services anytime and anywhere with guaranteed Quality of Service(QoS) and manageability by separating control functions from bearer and services. The Service Delivery Platform(SDP) provides common interfaces and protocols to deploy existing or new services in an efficient way. Therefore SDP over IMS plays a role of bridge between established network and new IMS network by simplifying the interaction among application services. In order to enrich the multimedia network communication, we try to deploy the Push-to-talk over Cellular(PoC) service which is considered as the outstanding and distinguished half-duplex Voice over IP(VoIP) application service among deployable candidate services over mobile network. In this paper we investigate the advantages of PoC service and PoC architecture firstly, and then focus on the its practical implementation for the prototype to validate the feasibility of its deployment and realization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8B
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• pp.797-804
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• 2009

Recently, SIP(Session Initiation Protocol) is used to set up and manage sessions for multimedia applications such as VoIP(Voice over IP) and IMS(IP Multimedia Subsystem). However, because SIP operates over the Internet, it is exposed to pre-existed internet security threats such as service degradation or service disruptions. Multimedia applications which are delay sensitive even suffers more from the threats mentioned above. The proposed methods so far to detect SIP INVITE flooding are CUSUM(Cumulative Sum), Hellinger distance and adaptive threshold, but among methods only take normal state into consideration. So, it is not capable of adapting the condition of the network congestion which are dynamically changing. In this paper, SIP INVITE flooding detection algorithm considering network congestion which enables efficient detections of such attacks is proposed. The proposed algorithm is expected to detect other types of attacks such as BYE and CANCEL more precisely compared to other methods.

• IEMEK Journal of Embedded Systems and Applications
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• v.4 no.2
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• pp.79-83
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• 2009

Streaming Technology can support the real-time playback without downloading and storing multimedia data in local HDD. So, client browser or plug-in can represent multimedia data before the end of file transmission using streaming technology. Recently, the demand for efficient real-time playback and transmission of large amounts of multimedia data is growing rapidly. But most users' connections over network are not fast and stable enough to download large chunks of multimedia data. In this paper, we propose an intelligent IP streaming system based on personalized media service. The proposed IP streaming system enables users to get an intelligent recommendation of multimedia contents based on the user preference information stored on the streaming server or the home media server. The supposed intelligent IP streaming system consists of Server Metadata Agent, Pumping Server, Contents Storage Server, Client Metadata Agent and Streaming Player. And in order to implement the personalized media service, the user information, user preference information and client device information are managed under database concept. Moreover, users are assured of seamless access of streamed content event if they switch to another client device by implementing streaming system based on user identification and device information. We evaluate our approach with manufacturing home server system and simulation results.