• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.2
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• pp.93-100
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• 2009

As the Internet Protocol be widely/rapidly used in packet communication, common carriers are providing the multimedia service(Both direction real-time voice, video conference, remote educational etc.)on the Internet. Also the 070 VoIP (Voice over IP) service is provided by the carriers on the packer network. In order to offer VoIP service in Korea, common carrier has to acquire the optimum level for QoS(quality of service). In this paper, we study on CODEC quality to get a higher QoS for VoIP.

• Venture DIGEST
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• s.103
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• pp.8-11
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• 2007

인터넷전화(VoIP)는 "Voice over Internet Protocol"의 약자로 기존의 회선교환망(Circuit Network)이 아닌 인터넷망(IP Network)을 통해 패킷단위로 전송하여 통화권 구분없이 음성등을 송신하거나 수신하는 새로운 방식의 전화서비스이다.

• Journal of information and communication convergence engineering
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• v.7 no.3
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• pp.309-313
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• 2009

In this paper, end-to-end VoIP(Voice over Internet Protocol) performance is evaluated by simulation with NS-2 simulation tool. There are many results studied and published for VoIP performance over TCP/IP networks. But, almost all of them were focused on wired or wireless Internet environments. About MANET (Mobile Ad Hoc Network), VoIP is currently studying several points of research. In this paper, analysis of VoIP performance is done with focusing on the mobility of MANETs. MOS(Mean Opinion Score), network delay, packet loss rates are considered as end-to-end QoS performance parameters.

• Journal of Korean Society of Industrial and Systems Engineering
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• v.33 no.3
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• pp.137-145
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• 2010

The objective of this paper is to evaluate the equilibrium price between PSTN and VoIP telephony services in the case of non-linear utility function. Currently there are two types of wired phone services we are known PSTN (Public Switched Telephone Network) and VoIP (Voice over Internet Protocol). The PSTN telephony which provide high quality service and VoIP which provides relatively low quality service form a vertically differentiated oligopoly. Therefore, the evaluation of the equilibrium price between PSTN and VoIP services is very important to wired phone service providers. The equilibrium price depends on the state of the service cost function has been proved different value. This paper was evaluated each equilibrium price for the state of the linear cost function and non-linear cost function. Subsequently, this paper analyzed the demand of both services and the equilibrium profit which can maximize the profit of both service providers.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.445-447
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• 2003

최근 무선네트워크의 급속한 확대로 무선인터넷 접속 또한 크게 증가하고 있다. 무선인터넷을 이용하는 Voice over IP 서비스는 IP 기반의 인터넷과 셀룰러 네트워크를 합쳐 놓은 것과 유사하며 모바일 노드의 이동성 확보가 핵심 기술이다. 특히 VoIP 서비스 이용자는 지연이나 끊어짐에 대하여 매우 민감하므로 가급적 지연시간이 적은 핸드오프 기법이 필요하다. 본 연구에서는 무선네트워크를 이용하는 VoIP 서비스 프로토콜 중 하나인 SIP를 기반으로 이동성 관리를 위한 신호의 흐름을 도시하여 발생 가능한 지연의 구성요소를 분석하였으며 핸드오프 지연을 줄이기 위한 Prediction Shadow Registration을 제안하였다.

• 한국정보통신설비학회:학술대회논문집
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• 2009.08a
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• pp.91-96
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• 2009

In Public Switched Telephone Networks(PSTN) number portability is implemented by utilizing Intelligent Network(IN) functions for number mapping. And voice over IP(VoIP) and IP Multimedia Subsystem(IMS) networks can deploy number portability by using E.164 Number Mapping(ENUM). This paper discuss the possibility of using E.164 Number Mapping(ENUM) for number portability in voice over IP/IP Multimedia Subsystem and Public Switched Telephone Networks, eliminating the need for Number Portability Database(NPDB) for number portability routing data in Public Switched Telephone Networks.

• Information and Communications Magazine
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• v.21 no.4
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• pp.65-73
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• 2004

인터넷전화 서비스를 제공하기 위한 대표적인 기술이 VoIP(Voice over Internet Protocol)이다. VoIP 기술에 의한 전화 서비스는 기존의 PSTN(Public Switched Telephone Networks)서버에 비하여 경제적이고, 향후 멀티미디어 서비스 지원 등의 특징을 가지기 때문에 보급의 확산이 기대된다. VoIP 기술은 유선에서 뿐만 아니라, 무선에서도 VoIP 기술을 채택하여 유무선 통합의 핵심기술로서 IETF, ITU 등에서 작업이 추진되고 있다［1］. 이와 같이 VoIP 서비스의 확대가 예상됨에 따라 사용자의 인증, 메시지 보호 등 보안 서비스의 중요도가 증대되고 있다［2］.(중략)

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.2
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• pp.279-287
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• 2001

In this paper, we discuss the scale of subjective speech quality measurement over VoIP(Voice over IP) network which is a component of broadband networks. Objective parameters of multimedia services like PSNR or jitter can easily measured and defined, but these factors are not easily meet the user's perceptual recognition. We suggest the speech quality measurement scale through the subjective measurement for end-to-end speech quality composed of sender-side quality, transmission quality, receiver-side quality, which provide the degree of correctness of representation of speaker, the degree of impairment caused by various factors, the degree of recognition of processed speech, respectively. Also, we examined the proposed method and verify it's availability.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.5
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• pp.942-949
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• 2006

The VoIP(Voice over IP) has been worldwide used and already put to practical use in many fields. However, it is needed to ensure secret of VoIP call in a special situation. It is relatively difficult to eavesdrop the commonly used PSTN in that it is connected with 1:1 circuit. However, it is difficult to ensure the secret of call on Internet because many users can connect to the Internet at the same time. Therefore, this paper suggests a new model of Internet telephone for eavesdrop prevention enabling VoIP(using SIP protocol) to use the VPN protocol and establish the probability of practical use comparing it with 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.