• Journal of the Korea Convergence Society
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• v.8 no.11
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• pp.37-42
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• 2017

Over the next decade, there are estimated to be 250 billion IoT devices in the future, and a variety of new services and device markets are expected to be created via a new network connection. However, it is difficult to accommodate various sensors and IoT devices, such as product and installation environments, for existing IP/SIP/IMS cameras and video door phone. Additionally, recently, sensors and actuators that can accommodate IoT technology and standards are continually updated. In addition, this paper has developed an SIP/IMS intelligent gateway to flexibly accommodate IoT related equipments and to interface with ZigBee/Z-Wave/NFC endpoints in addition to the gateway basic functions. This intelligent gateway will contribute to the development of the IMS system to replace the VoIP system with the IMS equipment due to the growth of the IMS market.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.1C
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• pp.36-47
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• 2002

MGCP ia standard protocol by IFTF, which is designed to facilitate inter-working with other protocols. Since the nature of protocol, it can be known as a proper solution to integrate heterogeneous networks consisted of various protocols and systems. In this paper MGCP is used for controlling VoIP gateways from external call control elements(MGCs) and introduces a method to improve interoperability for various MGC in a gateway based on MGCP. It presents implementation issues to provide the interoperability for MGC dependent parts such as call flows of message encoding, and a system architecture to resolve the issues.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.3B
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• pp.398-407
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• 2010

Due to increase of user requirement for various traffics and the advance of network technology, each distinct network has converge into FMC(Fixed Mobile Convergence) networks. However, we need to research the performance analysis of VoIP(Voice over Internet Protocol) in the FMC network to provide QoE for the voice user of FMC network. Therefore, this paper introduces the scenario which is the situation of voice quality degradation when a user uses VoIP to communicate with other users in the FMC network. Especially, this paper presents scenario in terms of the component of the network and finds the improvement point of voice quality. In the simulation results, three improvement points of voice quality are found as following: voice quality degradation by packet loss in the physical layer of the HSDPA network, by utilizing GGSN without QoS parameter mapping mechanism which is gateway between 3GPP and IP backbone, and by using non-QoS AP in the WLAN network.

• Proceedings of the IEEK Conference
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• 2001.06c
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• pp.225-228
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• 2001

There are two standards currently compete for the dominance of IP telephony architecture. Those are H.323 protocol suit by International Telecommunication Union Sector T(ITU-T) and Session Initiation Protocol/Session Description Protocol(SIP/SDP) by International Engineering Task Force(IETF). This paper has been studied a adaption VoIP in home networking, Especially SIP-based call screening service in home gateway. And then this paper has designed SIP-based call screening service in home gateway working protocol, verified them.

• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.172-175
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• 2003

In this paper we propose a method fur the evaluation of the quality of service for VoIP services in NGN. Specifically, let us anatomize the elements of delay of a voice connection in the network in an end-to-end manner and investigate expected value at each point. We extract the delay time in each element in the network such as gateway, network node, and terminal equipment, and estimate an upper bound fur the tolerable delay in each element.

• The KIPS Transactions:PartC
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• v.12C no.2 s.98
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• pp.261-266
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• 2005

This paper proposes two systems. One is a Home Gateway(HG) which efficiently connects and controls digital appliances in the home network environments. The other is a Management Server(MS) that overcomes the physical limitation of the HG. The HG supports networking modules(TCP/IP for Ethernet, ADSL), home networking functions(HomePNA, IEEE1394 PLC) and telecommunication system(PSTN/SLT, VoIP, Video Communication). The HG is expected to be a core device for the integrated digital home environments. The MS is a dedicated server which manages and controls individual HG, home appliances and HA devices implemented at an area.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.12
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• pp.2301-2306
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• 2008

SIP(Session Initiation Protocol) has become an universal standard for multimedia communications for both wired and wireless networks since it has been adopted as a standard protocol for IMS platform in 3GPP standardization organization at November 2000. In this paper, we design and implement a SIP-based softphone client program which provides telephony service between internet users and a call center equipped with VoIP gateway. A softphone client based on PC-to-phone connection should guarantee to provide interoperability with various VoIP gateways and higher portability to be able to operate on different PC environments. The softphone client program in this paper has been developed with SIP 2.0 standard protocol to support interoperability and with JAIN SIP and JMF package to achieve higher portability.

• 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.

• Proceedings of the Korean Information Science Society Conference
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• 2004.04a
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• pp.787-789
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• 2004

MEGACO(MEdia GAteway COntro) 프로토콜은 VoIP(Voice over IP) 시스템에서 MGC(Media Gateway Controller)와 MG(Media Gateway)간에 통신을 정의하는 표준이다. MEGACO 명세서에는 통신규약에 대한 내용을 ABNF (Argumented BNF)형식으로 제공하고 있으나, 이것을 그대로 사용하여 MEGACO 메시지를 분석하는 파서(Parser)를 개발하기에는 많은 어려움이 있다. 규칙(rule)과 규칙간의 비결정적인 요소와 토큰과 토큰간의 모호성이 많이 존재하기 때문에 적절한 변환을 통하여 파서를 제작하여야 한다. 본 논문에서는 ANTLR 파서 생성기와 MEGACO 명세서에서 제공되는 ABNF문법을 사용하여 MEGACO 파서(Parser)를 개발하였다. ANTLR에서 제공하는 Syntactic predicate와 Semantic Predicate등을 적절하게 사용하여 명세서에 존재하는 여러 가지 형태의 비결정적인 구문과 모호한 토큰들을 제거하였다.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.113-118
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• 2005

Softswitch is the core BcN equipment which voice and multimedia switching based on the IP Technologies. It is designed to replace the Class 5(local Exchange) and Class 4(Toll Exchange) switch based on the circuit wired and wireless switching network technologies. Softswitch gets its name because typically it is a software based solution implemented on general purpose computers/servers. While the traditional PSTN switches are rely on dedicated facilities for T and S inter-connection and are designed primarily for voice communications. Packet based Softswitch is divided the control of call and bearer, very different from Public telephone network. Sometimes Call Agent or Media Gateway Controller, a key component in the VoIP solution, is also called Softswitch. This paper will suggest the software architecture of softswitch for performance in call processing part, also suggest the session management model to cover call setup latency.