• Proceedings of the Korea Information Processing Society Conference
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• 2011.04a
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• pp.46-49
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• 2011

This paper is to implement MoIP platform to send and receive video and audio at the same time by using high-performance Dual Core Processor. Even if Wall-Pad key component of a home network system is released by using embedded processors, it's lacking of performance in terms of multimedia processing and feature of video telephony through which video and voice are exchanged simultaneously. The main reason could be that embedded processors currently being used do not provide enough performance to support both MoIP call features and various home network features simultaneously. In order to solve these problems, Dual processor could be used, but in the other hands it brings another disadvantage of high cost. Therefore, this study is to solve the home automation features and video telephony features by using Dual Core Processor based on ARM 11 Processor and implement the MoIP Wall-Pad which can reduce the board design costs and component costs, and improve performance. The platform designed and implemented in this paper verified performance of MoIP to exchange the video and voice at the same time under the situation of Ethernet network.

• 한국HCI학회:학술대회논문집
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• 2008.02b
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• pp.317-322
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• 2008

WCDMA 의 등장으로 영상통화를 지원하는 휴대폰의 보급은 빠르게 확산되고 있다. 하지만 예상과 달리 사용자들의 실제 영상통화 이용율은 그리 활성화 되지 못하고 있는 실정이다. 본 연구자들은 이러한 현상이 나타나게 된 원인이 영상통화 기능이 단순한 통화기능으로만 활용될 뿐 그 실질적 효용성이 높지 못하기 때문이라 판단하여, 영상 통화를 활성화 시킬 수 있는 방안에 대한 연구를 진행하였다. 오늘날 IT 기반의 서비스 중에 매우 활성화 된 것이 쇼핑 관련 서비스이다. 무선 인터넷과 모바일 환경에서 금융과 통신의 융합이 사회적 화두로 떠오르고, 이러한 이유로 영상통화 기능을 쇼핑과 연계한다면 그 효용성이 매우 커질 것으로 판단하였다. 그리하여 본 연구에서는 사용자 행태 조사을 통해 그들의 needs 를 범주화하고 분석하여 영상통화 쇼핑 서비스 어플리케이션을 도출하였다. 그리고 도출된 어플리케이션의 세부적 절차와 원활한 인터페이스의 설계를 위해 시나리오 형식으로 개방하여 그 형태를 구체화시키고자 하였다.

• Proceedings of the KIEE Conference
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• 1998.07g
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• pp.2414-2416
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• 1998

This paper presents real-time implementation of speech vocoder for PSTN video telephony using ITU G.723 16Kbps ADPCM algorithm. The ADPCM encoder accepts 8-bit PCM compressed signals and expends it to a 14-bit-per-sample. The predicted values are subtracted from encoded signals to produce difference signals. Adaptive quantization is performed on the difference signal to produce a 2-bit, output for transmission over the channel. Computer simulations and experiments were performed to evaluate the performance of the speech vocoder.

• The Journal of Korea Robotics Society
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• v.5 no.2
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• pp.93-101
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• 2010

This paper presents the system design and implementation of 'Mon-e', the KT's edutainment robot for young children. We paid our special attention to the computer- illiterate young children, and the provision of the physical and friendly human interface of robots. Specifically, the paper focuses on the video telephony and home monitoring service using the Mon-e robot. RFID cards -based calling makes it possible for the computer-illiterate children to make a phone call to their parents. The SIP and DTMF based remote control of the robot enables the search and track of the children. This experimental development shows the potentialities and values of the convergence service of telecommunication and robotics.

• The KIPS Transactions:PartD
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• v.13D no.3 s.106
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• pp.383-390
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• 2006

Product-line engineering is a dreaming goal in software engineering research. Unfortunately, the current underlying technologies do not seem to be still not much matured enough to make it viable in the industry. Based on our experiences in working on mobile telephony systems over 3 years, now we are in the course of developing an approach to product-line engineering for mobile telephony system software. In this paper, the experiences are shared together with our research motivation and idea. Consequently, we propose an approach to building and maintaining telephony application logics from the perspective of scenes. As a Domain-Specific Language(DSL), Menu Navigation Viewpoint(MNV) DSL is designed to deal with the problem domain of telephony applications. The functional requirements on how a set of telephony application logics are configured can be so various depending on manufacturer, product concept, service carrier, and so on. However, there is a commonality that all of the currently used telephony application logics can be generally described from the point of user's view, with a set of functional features that can be combinatorially synthesized from typical telephony services(i.e. voice/video telephony, CBS/SMS/MMS, address book, data connection, camera/multimedia, web browsing, etc.), and their possible connectivity. MNV DSL description acts as a backbone software architecture based on which the other types of telephony application logics are placed and aligned to work together globally.

• Journal of Digital Convergence
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• v.12 no.8
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• pp.235-242
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• 2014

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. In this paper we suggested an Asterisk-based Internet telephony system which can be easily scalable. Most current systems use text files to manage their configuration: SIP users, dialplans, IVR service and etc. But we designed the management system which introduces database tables for efficiency and scalability. It also supports web-based functions developed by using Asterisk, Apache, MySQL, jQuery, PHP and open source softwares.

• Journal of Digital Convergence
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• v.10 no.6
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• pp.259-267
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• 2012

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. Recently, the introduction of smart phones has led to a growth in social networking services and thus, the research and development of Internet telephony has been actively progressed and has the potential to become a replacement for the telephone service that is currently being used. In this paper we designed and implemented an Internet telephony network system which is developed by using Asterisk and open source softwares. It is developed on the linux system and has some features such as VoIP telephony service between SIP phones, voice mail, and call recording. It also supports web-based functions such as SIP users and server system management that is implemented by Apache web server and PHP programs. Afterwards, this system will be applied as VoIP network base technology for small sized companies and organizations. It will paly a role for encouraging companies to use open source softwares.

• 한국정보통신설비학회:학술대회논문집
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• 2009.08a
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• pp.99-103
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• 2009

Recently, as wired and wireless communication services have rapidly developed and multimodal mobile devices which have various characteristics have widely spread, the need for new convergence services increases. The growing population of VoIP technologies and the high communication expense yield that the market of IP based telephony such as WiFi phone and IP phone is substituted for one of the conventional PSTN telephony. With the help of this trend, the wireline network operators desire to find a market in mobile networks. Therefore, they focus on Fixed Mobile Convergence (FMC) service as one of the key factors to accomplish this goal. FMC services are able to provide the mobility of voice services between circuit switched and packet switched networks. IP Multimedia Subsystem (IMS) based Voice Call Continuity (VCC) is one of the schemes to embody FMC services. As Application Server (AS) which has this VCC function provides seamless handover of services between heterogeneous networks, FMC subscribers can communicate seamlessly with others m WiFi domain and COMA domain using WiFi-COMA dual phone. Most of enterprises have already introduced IP network infrastructure and IP-PBX (Private Branch eXchange) for telephony. However, the problems of high communication cost and work inefficiency due to frequent outside jobs or business trips have remained. In order to solve these problems, demands for enterprise FMC services increase. In this paper, we introduce a new IP-PBX based VCC model that can provide seamless handover of voice services between WiFi and COMA networks for enterprise users and we investigate some interworking and security issues between Soft Switch (SSW) and IMS, or between IMSs. In addition, we introduce a new service that can provide the continuity of voice sessions as well as video sessions using Multimedia Session Continuity (MMSC) technology which has evolved from VCC. This service is expected to be one of the next-generation personalized services based on user's context.

• Journal of Korea Society of Industrial Information Systems
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• v.24 no.5
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• pp.105-113
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• 2019

Korean military uses 'Tactical information communication network' to exchange information between units. In this study, we developed a media processing board for multi-party voice and video telephony based on open source software. On the other hand, in order to apply open source software for weapon systems and parts that are mounted on weapon systems, appropriate review is required according to the weapon system software development and management manual of the Defense Acquisition Program Administration (DAPA). In this study, the analysis of the requirement items was performed and the appropriate countermeasures were proposed for the open software applied to the media processing board with respect to 'the guidelines for the application of weapon systems to open source software', an appendix to the DAPA's manual.

• Proceedings of the IEEK Conference
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• 2000.06a
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• pp.225-228
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• 2000

In this paper, a design of the reliable video-conference system using CTI(Computer Telephony Integration) technology is proposed. When video-conference is run on the current existing Internet, the transmission delay problem for voice data traffic can be frequently occurred. In order to transmit the real-time voice data through the Internet efficiently, some complicated algorithms such as CODEC(Code/Decode) should be applied. It can cause further excessive processing delay which can affect the overall performance. The voice traffic is usually transmitted through the reliable PSTN(Public Switched Telephone Network) in the CTI system. In this paper a new architecture, in which PSTN for voice traffic and Internet for video traffic are used at the same time instead of using Internet by itself, is proposed to relieve the problems on a video conference.