• Journal of Information Technology Applications and Management
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• v.17 no.2
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• pp.133-154
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• 2010

Recently, Internet Telephony has become increasingly popular in telecommunication industry. However, previous research on Internet Telephony has focused on analyzing specific Internet Telephonysolutions, identifyingthe Internet Telephony movement itself. The research on prediction models about Internet Telephony adoption has been minimal. The main propose of this study is to develop models for predicting transition intention from using traditional telephones to using Internet Telephony. To do so, this study uses data mining methods to analyze demands in the IT communications market and to provide management strategies for Internet telephony providers. Especially this study uses discriminant analysis, logistic regression, classification tree, and neural nets to develop those prediction models toward Internet Telephony adoption. The models are compared with each other and a superior model is chosen.

• Proceedings of the Korea Database Society Conference
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• 2010.06a
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• pp.187-199
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• 2010

Recently, Internet Telephony has become increasingly popular in telecommunication industry. However, previous research on Internet Telephony has focused on analyzing specific Internet Telephony solutions, identifying the Internet Telephony movement itself. The research on prediction models about Internet Telephony adoption has been minimal. The main propose of this study is to develop models for predicting transition intention from using traditional telephones to using Internet Telephony. To do so, this study uses data mining methods to analyze demands in the IT communications market and to provide management strategies for Internet telephony providers. Especially this study uses discriminant analysis, logistic regression, classification tree, and neural nets to develop the prediction models for the Internet Telephony adoption. The models are compared with each other and a superior model is chosen.

• Proceedings of the IEEK Conference
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• 2000.07b
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• pp.691-694
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• 2000

This paper is addressed to Internet telephony based service implementation. It describes an implementation method which uses ARS as a gateway which combines Internet and traditional public switched telephone network for Internet information service using telephony and fax, is different to traditional Internet telephony which provide enhanced speech quality and low cost functionality. This method allows telephony and/or fax user to get Internet information without additional Internet bill, Internet infrastructure and low connection quality from low signal bandwidth connected him. Implemented system is useful to a special kind information service such as climate information of Korea etc and simpler than WAP based service as for wireless mobile telephony user. We implement job opportunity information and advertisement service supported by Home page of Choong Book Small & Medium Business Administration and e-mail service supported by Korean Society for Rehabilitation of persons with Disabilities to demonstrate the system ability. As a result of test implementation, this service system works good fur blind persons and graduated persons without job, is expected to apply for special Internet information provider via Voice and Fax.

• 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.

• Journal of Industrial Technology
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• v.26 no.B
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• pp.199-206
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• 2006

Internet telephony provides voice communication services with added flexibility for multimedia extension at a lower cost compared to traditional telephone systems. We implemented an internet telephony system as an overlay network without a centralized server, using a distributed hash table (DHT). Compared to the current server-based internet telephony system, our system is fault-tolerant, scalable, and can be flexible extended to various services and advanced to integrated service. To demonstrate the high flexibility of our DHT-based internet telephony system, we made our system cooperate with web servers. Web users can check up others' online stales and establish voice communication sessions to online users at a mouse click. This technology can be applied to more complex services such as multimedia messaging or video conference service.

• 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.

• Journal of KIISE:Computing Practices and Letters
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• v.9 no.1
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• pp.47-59
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• 2003

The Internet telephony service is one of the commercially successful Internet application services. VoIP technology makes the service come true. VoIP deploys H.323 or SIP as the standard protocol for the distributed multimedia services over the Internet in which QoS is not guaranteed. VoIP carries the packetized voice over the RTP/UDP/IP protocol stack. The data transmission trouble is caused by UDP when the service is provided in private networks and some ISP-provided Internet access networks in the private address space. The Internet telephony users in such networks cannot listen the voices of the other parties in the public Internet or PSTN. Making the problem more difficult, the Internet telephony service considered in this paper gets the incoming voice packets of every session through only one UDP port number. In this paper, three schemes including the terminal proxy, the gateway proxy, and the protocol translation are suggested to solve the problems. The design and implementation of the NAT proxy server based on gateway proxy scheme are described in detail.

• 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 the Korea Safety Management & Science
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• v.11 no.3
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• pp.199-207
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• 2009

IP Telephony service was restricted to an outgoing call and low quality since the trust domestic IP Telephony service launch of Saerome co. Ltd, in Jan. 2000. However, Interest of IP Telephony service, which is substituted for PSTN, has been highly elated because of the developed equipment softswitch and new technology. This kind of importance and marketing of VoIP are recognized to telecommunication providers. With this trend, they try to administrate customer satisfaction and invest R&D to survive in this hard competition and unexpected change. To achieve this objective, they should try to realize the searching process of the quality decision attribution (QDA). However, there is little research on the aspect of service quality of Internet telephony so far. For this, the investigator established the tangibles, the reliability, the responsibility, the assurance, the empathy, the charge with information sources as core elements. In order to examine the influence of IP Telephony service upon the attitudes toward a brand and the purchase intention.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.6 no.3
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• pp.386-392
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• 2002

Internet telephony is an application that transmits voice data for conversation. Therefore it must provide high sound quality. However while audio packets are transferred through the network, they are affected by delay variations and jitters, which could result in poor sound quality of the receiving end does not have an appropriate jitter buffer to overcome network factors. This thesis introduces a buffer management algorithm that could be used to provide better sound quality for Internet phone terminals. This algorithm actively responds to both the compression algorithms that are used by the terminals, as well as to the received data to provide an improvement in sound quality. In order to verify the effectiveness of the proposed algorithm, we experimented in variance network settings. The results show that the proposed algorithm improves on the performance of the conventional buffer management algorithm.