• Journal of Digital Convergence
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• v.16 no.2
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• pp.9-17
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• 2018

The purpose of this study is to find out how voice recognition technology can be utilized to solve the elderly problem rapidly aging in Korea. Public support services and civilian nursing services for the elderly are expected to expand in Korea. In this case, voice recognition technology can be used variously for the elderly who are not familiar with the media interface. To this end, our researchers visited Japan and examined the achievements obtained by voice recognition technology in the elderly care. Especially, when caregivers write reports, they have greatly reduced their working hours by replacing the handwritten reports with ones using voice recognition technology. This method can be easily implemented in Korea. In addition, the social cost of the elderly support can be gradually reduced through the development of a robot equipped with voice recognition technology. Consequently, we realize that when voice recognition technology is combined with artificial intelligence programs of various emotion recognition functions and various policy possibilities as well.

• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.67-74
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• 2002

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rate to send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link, An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used for feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• 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.

• 한국IT서비스학회:학술대회논문집
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• 2002.06a
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• pp.268-274
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• 2002

현재 대부분의 대학 입시정보시스템은 ARS 및 웹기반의 서비스를 병행하여 제공하고 있다. 기존 ARS 기반 시스템의 단점은 전화버튼만으로 입력이 제한된다는 점과 시스템의 구축 및 유지보수가 용이하지 않다는 점이다. 이러한 문제점을 해결하기 위하여 전화버튼뿐만 아니라 음성인식에 의한 입력이 가능한 VoiceXML 음성 애플리케이션을 도입하였다. VoiceXML 및 음성 애플리케이션을 활용하여 입시정보시스템을 설계 및 구현해 본 결과 이러한 문제점들을 상당부분 해결할 수 있었다. 그리고 미리 연관된 키워드를 등록하여 다양한 입력옵션을 제공함으로써 자연어 처리가 좀더 용이해졌다. 이는 XML의 최대장점인 다양한 확장성과 응용성이 향상되는 것이며 사용자에게 기존 시스템보다 훨씬 개선된 사용자 인터페이스를 제공할 수 있게 된 것이다. 또한 기존 웹기반의 서비스에 쉽게 연동이 가능하고 유지보수 또한 기존 시스템보다 쉽게 할 수 있다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.6
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• pp.89-94
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• 2015

FMS which has a concept that wireless terminal can replace wire terminal services is a technologies that is can provide service costs same as wire terminal in the special zone. Enterprise type of FMC that is developed making up for the weak point is must have to improve voice quality and FMC performance in the soft phone. This paper measure voice quality based on the one way of the total estimated delay time of FMC to carry out IMS services between IP-PBX and FMC soft-phone to operate it's controller optimally and put forward evidence to be in 120ms and 150ms in the VoIP FMC voice quality. To measure FMC performances in four categories evaluated trials and prove its performances.

• Journal of Advanced Marine Engineering and Technology
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• v.27 no.1
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• pp.65-75
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• 2003

The internet telephony service is one of the successful internet application services. VoIP is the key technology for the service to come true. VoIP uses H.323 or SIP as the standard protocol for the distributed multimedia services over the internet environment, in which QoS is not guaranteed. VoIP carries the packetized voice by using the RTP/UDP/IP protocol stack. The UDP-based internet services cause the data transmission problem to the users behind the internet firewall. So does the internet telephony service. The users are not able to listen the voices of the counter-parts on the public internet or PSTN. It makes the problem more difficult that the internet telephony service addressed in this paper uses only one UDP port number to send the voice data of all sessions from gateway to terminal node. In this paper, two schemes including the usage of dummy UDP datagrams, and the protocol conversion are suggested. The implementation of one of the schemes, the protocol conversion, and the performance evaluation are described in detail.

• The Journal of the Acoustical Society of Korea
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• v.28 no.2E
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• pp.35-42
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• 2009

In this paper, we present criteria of objective measurement of speech quality to provide the mobile-VoIP services efficiently over wireless mobile internet. The mobile-VoIP service, which is based on mobility and is error-prone compared to conventional VoIP over wired network, is about to be launched, but there have not been adequate quality indexes and the Quality of Service (QoS) standards for evaluating speech quality of Mobile-VoIP. In addition, there are many factors influencing on the speech quality in packet network of which packet loss contribute directly to the overall voice communication quality. For this reason, we adopt the Gilbert-Elliot Channel Model for modeling packet network based on IP and assess the voice quality through the objective speech method of ITU-T P. 862 PESQ and ITU-T P. 862.1 MOS-LQO under various packet loss rates in the transmission channel environments. Our simulation results address the specific criteria and QoS for the mobile-VoIP services in terms of the various packet loss environments.

• Journal of KIISE:Computing Practices and Letters
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• v.15 no.8
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• pp.582-585
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• 2009

VoIP technology for transmitting voice over IP network such as packet-based network has a lot of benefits by integrating services and reducing costs. The network is different from PSTN-based communications in some aspect such as transmitting not only voice but also text, image, multimedia data. In addition, portable terminals like a mobile phone, and ubiquitous communicator can easily access the internet for VoIP. Therefore, To prevent illegal users, offering certificate services is necessary, This study proposes a solution of user certification for a VoIP environment.

• 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.

• The Transactions of the Korea Information Processing Society
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• v.7 no.4
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• pp.1211-1216
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• 2000

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rateot send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link. An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used fro feedback control. Simulation results are presented, which show the effectiveness of the algorithm.