• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.6
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• pp.63-69
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• 2010

Today, Internet Telephony Services based on VoIP are gaining tremendous popularity for general user. Therefore a various demands of the user keep up increase, the most important requirements of these is voice security about telephony system. It is needed to ensure secret of voice call in a special situation. Due to the fact that many users can connect to the internet at the same time, VoIP can always be in a defenseless state by hackers. Therefore, in this paper, we have developed VPN IP-PBX for the voice security and measured conversation quality by adopting VPN IPsec based on SIP and using tunnel method in transmitting voice data to prevent eavesdrop of voice data. This VPN IP-PBX that is connected Soft-phone provide various optional services.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.2
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• pp.73-80
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• 2009

Although it is relatively difficult to eavesdrop the commonly used PSTN in that it is connected with direct circuit, it is difficult to ensure the secret of call on Internet because many users can connect to the Internet at the same time. However, it is needed to ensure secret of voice call in a special situation. Due to the fact that many users can connect to the internet at the same time, VoIP can always be in a defenseless state by hackers. Therefore, in this paper, we have developed the increased voice security internet telephone terminal and measured conversation quality by adopting VPN PPTP based on SIP and using tunnel method in transmitting voice data to prevent eavesdrop of internet telephone.

• Journal of Digital Convergence
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• v.9 no.5
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• pp.225-233
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• 2011

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 a recording system which records voice data of SIP-based Internet telephone's voice calls. It is developed on the linux system and has some features such as audio mixing of two in/out voice channels, live packet sniffing, and the ability to transfer mixed audio files to the log file server. These functions are implemented using various open source softwares. Afterwards, this VoIP recording system will be applied as a base technology to advanced services like a VoIP-based call center system.

• Journal of Internet Computing and Services
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• v.5 no.6
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• pp.31-43
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• 2004

Voice over Internet Protocol (VoIP) is currently a popular research topic as a real time voice packet transmission method. But current Internet environment do not guarantee the quality of voice when we take a side view of delay, jitter and loss. Up to now, many voice based evaluation algorithms have been used to measure speech quality of VoIP systems. However, these algorithms have the defects that their results are different according to voice samples and some algorithms can not take network environment for speech transmission path. The E-model can be used to solve the problems of these algorithms. In this paper. we introduce VoIP planning guidelines through the various analysis of E-model which can model impairments of network quality as well as VoIP equipment quality systematically, We, also, show the evaluation method and results of speech transmission quality.

• Journal of Advanced Navigation Technology
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• v.21 no.1
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• pp.113-118
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• 2017

As interest in mobile devices and networks has increased recently, voice over internet protocol (VoIP) service, which is a technology for transmitting voice data using an existing internet protocol (IP) network, has rapidly spread, Cheap voice call service has become possible. As the digital broadcasting service becomes popular, hybrid fiber coaxial (HFC) network technology, which uses broadband cable network through fusion of broadcasting and communication, utilizes existing communication system and network equipment to provide various new services such as interactive broadcasting service. Therefore, if UGS-AD is applied to VoCM and RTPS is applied to MTA in order to guarantee the quality of voice data in actual HFC Internet service network, it is possible to smoothly perform voice data transmission in narrow upstream band which is a problem in actual commercial HFC network We also proposed a method to improve VoIP service by improving QoS of voice data in HFC Internet service network.

• The Journal of the Acoustical Society of Korea
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• v.37 no.6
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• pp.512-517
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• 2018

Improving voice quality is a major concern in telecommunications. In this paper, we propose a robust speech quality enhancement against background noise and packet loss at VoIP (Voice-over-IP) receiver. The proposed method combines network jitter estimation based on hybrid Markov chain, adaptive playout scheduling using the estimated jitter, and speech enhancement based on restoration of amplitude and phase to enhance the quality of the speech signal arriving at the VoIP receiver over IP network. The experimental results show that the proposed method removes the background noise added to the speech signal before encoding at the sender side and provides the enhanced speech quality in an unstable network environment.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.6
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• pp.17-22
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• 2013

Heterogeneous radios are used in disaster management agencies for a variety of reasons though the radio must have the same radio frequency and protocol for voice communication. For this reason, the variety of heterogeneous radio voice connection methods have been studied but these are simple analog voice line cross connection or partial networked based on digitalization. In this paper, we suggest the method of voice packet transmission method based on All-IP per radio through IP network using SIP/RTP for scalability and openness and developed a prototype of the proposed method was verified.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.4
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• pp.115-122
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• 2009

These day, market demanded to a Video conference systems rapidly increases in our life for cut cost in communication. more and more it will be grow up. but the cost building to a Voice conference and a Video conference is very hight. therefore it is builded around the big company and the public office. so in this study, we have developed to hybrid IP-PBX which is able to a Voice conference and a Video conference with one system. the system developed has the merits to low-price for it's building in a small company. we make proof the performance through the test. with using the hybrid IP-PBX, we can sharply reduce to communication cost.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.5
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• pp.95-100
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• 2011

The next generation IP-PBX system are demanding multimedia facility to carry out UC(Unified Communication) and voice security also. Therefore, in this paper, we have developed the integrated solution of IP-PBX for the voice security by adopting VPN IPsec based on SIP using tunnel method in transmitting voice data to prevent eavesdrop of voice data and have shared between communication system based on PC and PSTN terminals. In particular, We have developed a video conference, private switching, distributed processing and measured telephone conversation quality. This IP-PBX that is connected Soft-phone provide various optional services.

• 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.