• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.4
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• pp.39-44
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• 2014

IMS(IP Multi-Media Subsystem) base on the IP service platform which can offer multimedia as the voice, audio, video, and data is service platform. In 3G mobile communication in the early day, IMS had a suggestion for supporting to multimedia service in the 3GPP. But now It is broadly substituting in the IPTV, wire phone company and it is substituted in internet platform base on the soft-switch in currently. Especially nowadays, 4G LTE in a mobile communication company is rapidly growing in market. Therefore, in this study, we had designed to the main prosser that can admit to 1,000 user over and SIP gateway which can link the IMS 코어 that can link SIP Device which adopt the standard protocol on the SIP.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.2
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• pp.103-111
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• 2010

For that reason, more and more growing of the internet speed, many kind of the multimedia as if the voice, a letter, a photograph, a moving picture are appearance and integration of the internet as if the wireless internet, the CATV, a satellite is forming. Today, many changes in the communication environment are occurrence. therefore a demands of the user keep up increase. so, in this paper, we developed the SOHO IP-PBX embedded which can automatically exchange the call. the developed IP-PBX have many functions as if the exchange call function, the gateway function, the PSTN backup function, the FAX function, the VPN function, etc. and verified it by an experiment on the rules in KT.

• Proceedings of the Korea Information Processing Society Conference
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• 2007.05a
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• pp.1221-1224
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• 2007

현재 기업들 간의 전화비를 줄이고 각종 VoIP 부가 서비스를 위해 VoIP 시스템의 도입이 시작 되었다. 이에 VoIP 전화기들 간의 각 기능을 최대한 활용하기 위해 현재 VoIP 시장의 90%를 차지하고 있는 H.323 게이트웨이(Gateway)간의 H.323 프로토콜의 구현차이로 인한 문제점이 발생되기 시작되었다. 본 논문은 VoIP Gateway상에 H.323 프로토콜 통신을 하면서 프로토콜 연결 상 구현의 차이로 인해 VoIP 서비스에 비정상적인 작동으로 호가 종료가 되거나 음성이 들리지 않는 현상 및 전화기의 부가서비스를 사용할 수 없는 문제를 해결하기 위해 H.323 프로토콜의 작동을 분석하고 이기종간의 H.323프로토콜 신호가 호환되지 않을 경우 이를 해결하기 위해 H.323 프로토콜상의 H.245 시그널링 (signaling)을 Media gateway 서버를 이용해 구현한 RFC 2833 DTMF-compliant 프로토콜을 사용하여 H.323 프로토콜 처리함으로써 이기종간의 Call transfer, Hold 그리고 Conferenct 기능에 대한 호환성이 개선됨을 보여 준다.

• The KIPS Transactions:PartC
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• v.11C no.6 s.95
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• pp.799-806
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• 2004

A growing diversity of pervasive devices is gaming access to the Internet and other information. However, much of the rich multimedia contents cannot be easily handled by the client devices because of the limited communication, processing, storage and display capabilities. The in-tegration of voice, data and video service modified the target of networking technologies. Networks must have some the capabilities for in-tegration of various services and also for QoS support as required by each of those service. Because of these reasons, we developed EAGIS(Efficient ATM Gateway for real time Internet Service) to provide seamless multimedia service between the ATM network and the Internet. EAGIS consists of the interworking unit, content server, transcoding server, and the serveice broker to provide seamless multimedia service be-tween the ATM network and the Internet. In this paper, we design the architecture and transcoding service scenario of the EAGIS. When the RTP is used for the bi-directional communication, transcoding time is configured by the time-stamp of RTCP. When HTTP is used for unidirec-tional communication, self-timer is used. By using these reference time, standard transcoding method is applicable according to the frame trans-mission rate and network traffic load. And we can also assure the QoS of the multiple users` effective bandwidth by our algorithm.

• Journal of Advanced Marine Engineering and Technology
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• v.39 no.7
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• pp.744-753
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• 2015

This paper presents the results of research on developing marine unified communications to provide VoIP service based on marine satellites. With the recent popularity of smart-phones and other mobile devices, the demand for Internet-based wired and wireless unified technology has been growing in marine environments, and increasing interest is being directed to VoIP products and service models with high price competitiveness and the ability to deliver a variety of services. In this regard, this research designed three instruments, developed their unit modules, and verified their performances. These three instruments included the following: (1) a marine VoIP module equipped with an analogue gateway that can be linked to the existing devices used in vessels, which is more than 80% smaller than that of a land system; (2) a text/voice/video engine for marine satellite communications that runs on technology that minimizes communication data usage, which is a core technology for a marine VoIP service; and (3) a unified communication service that can support multilateral cloud-based message conversations, telephone number-based call functions, and voice/video calling between a private space in a ship and shore.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.12B
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• pp.719-727
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• 2007

The Wireless Mesh Network technology is able to provide an infrastructure for isolated islands, in which it is difficult to install cables or wide area such as battlefield of armed forces. Therefore, Wireless Mesh Network is frequently used to satisfy needs for internet connection and active studies and research on them are in progress However, as a result of increase in number of hops under hop-by-hop communication environment has caused a significant decrease in throughput and an increase in delay. Considering the heavy traffic of real-time data, such as voice or moving pictures to adaptive WMN, in a military environment, it is restricted for remote units to have their Mesh Node to get real-time services. Such phenomenon might cause an issue in fairness. In order to resolve this issue, the Location-based Packet Scheduling Scheme, which can provide an fair QoS to each mesh node that is connected to each echelon's AP and operates based on WRR method that gives a priority to emergency message and control packet. The performance of this scheme is validated.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4A
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• pp.377-384
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• 2006

According to fast VoIP technology development, more and more companies change voice network into IP based network among branch offices. IP PBX, which is deployed up to now, composed of IP phone and VoIP Gateway. Every telphone has replaced with If phone which support VoIP and VoIP gateway is installed in PBTN connection point to relay voice data. It can reduce the communication expense of International call, long distance call and call between a headquater and a trance because it uses internet line. In this paper, IP PBX is implemented that can distribute call using PBX network only usig personal terminal without Proxy Server. Depending on Role, terminal can be registered Master, Server and Client and it is verified in terms of performance and validation.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.142-145
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• 2001

I did research or design technology of(integration) network of PSTN and PSDN in this paper. Network structure model in association with soft switching and access gateway can create various services based on web and live time and service fee at the rate of bandwidth which can't be provided by existing network based on SONET. Also, interns of capability(function) of each area and construction of local network subscribers, it has great recovery capability by separating each network as local areas. It indicates that it can additionally maximize the application of its own communication network and drastic reduction of communication cost because it is alternated to exclusive usage cost.

• The Journal of the Korea Contents Association
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• v.11 no.1
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• pp.34-41
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• 2011

At present wireless internet access service is available through the 3G network, mobile WiMAX and WiFi anytime and anywhere. In this environment where there are various networks, users should be able to select specific networks depending on different situations. And it is necessary to provide mobility support between homogeneous and between heterogenous networks. Given this situation, the many proposals have been presented to link 3G, which has the largest service area among various networks, with mobile WiMAX(IEEE 802.16e), or with WiFi(IEEE 802.11). But, recently, with the increasing volume of wireless internet use and wireless internet data, due to the advents of net-book, e-book and smart phone, the service area of WiFi and mobile WiMAX has rapidly expanded. Especially, the availability of real-time application such as internet phone has led to the relative shrinking of the proportion of 3G mobile communication network giving conventional voice service, and enlargement of those of wireless internet access networks like WiFi and mobile WiMAX. This paper suggests a handover scheme based on PMIPv6, whitch support mobility between WiFi and mobile WiMAX, and minimizes handover delay. In this scheme, the mobile node has a dual stack structure composed of two interfaces-WiFi and mobile WiMAX. Since WiFi dose not support mobility, it is suggested that the mobile node have the capacity to deal with handover signaling between gateway in case of handover between homogeneous networks. This handover scheme, suggested comparing with current handovers between homogeneous networks, has proved, in its analytic evaluation, to be able to reduce handover, transmission, and signaling overhead.