• 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.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.3
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• pp.309-316
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• 2016

Recently, the voice service over a mobile communication system tends to be provided based on the packet-based technology. Even though the sufficient transmission rate is supported by LTE mobile communication system, the quality of VoIP service that is experienced by the user can be degraded by the change in the transmission conditions and the terminal mobility. This paper has established an environment on which experiments are conducted for the different values of the major parameters that represent the transmission conditions. The result can contribute to the decision of the requirement that the mobile system should meet for maintaining the quality of VoIP service.

• Annual Conference of KIPS
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• 2006.05a
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• pp.1243-1246
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• 2006

초고속 통신망의 확대 적용으로 인터넷의 빠른 성장과 함께 음성과 비디오 그리고 데이터를 통합하고자 하는 노력이 시도되고 있다. VoIP(Voice over IP)는 IP를 이용하여 음성과 데이터를 패킷 형태로 통합하여 실시간으로 전송하는 기술이다[1]. 패킷 네트워크에서 VoIP 시그널링 기술을 이용하면 망 자원으 효율적 이용 및 PSTN에 가까운 음질 그리고 인터넷과 연계한 다양한 음성서비스 지원이 가능하다. 콜센터에서도 VoIP를 사용하게 됨에 따라 VoIP망에서의 녹취 시스템이 필요하다. VoIP 녹취 시스템은 상담원과 고객 간의 통화 내용을 자동으로 녹음하여 보관함으로써 고객의 요구사항을 명확하게 파악할 수 있으며 녹취 데이터의 통계 자료 제공으로 효율적인 관리가 가능하고, 선택 녹취, 스케쥴링 녹취, 상담원의 평가 자료를 제공하여 고객 관리의 질적인 향상을 지원한다. 본 논문에서는 성능에 큰 영향을 주지 않고 기존의 VoIP 녹취 시스템의 문제점을 해결한 에이전트를 포함한 VoIP 녹취 시스템을 제안한다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.4
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• pp.737-744
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• 2012

Recently VoIP has provided voice(both wired and wireless from IP-based) as well as the transmission of multimedia information. VoIP used All-IP type, Gateway type, mVoIP etc. Wired and wireless VoIP has security vulnerabilities that VoIP call control signals, illegal eavesdropping, service misuse attacks, denial of service attack, as well as wireless vulnerabilities etc. from WiFi Zone. Therefore, the analysis of security vulnerabilities in wired and wireless VoIP and hacking incidents on security measures for research and study is needed. In this paper, VoIP (All-IP type, and for Gateway type) for system and network scanning, and, IP Phone to get the information and analysis of the vulnerability. All-IP type and Gateway type discovered about the vulnerability of VoIP hacking attacks (Denial of Service attacks, VoIP spam attacks) is carried out. And that is a real VoIP system installed and operated in the field of security measures through research and analysis is proposed.

• Proceedings of the KIEE Conference
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• 2001.11c
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• pp.130-133
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• 2001

VoIP(voice over internet protocol) technology is based on IP protocol. The IP protocol can be involved in two types of communication: unicasting and multicasting. Unicasting is the communication between one sender and one receiver. It is one-to-one communication. Multicasting is one-to-many communication. So that, many receivers can get same data from one sender simultaneously. and, the different protocol are proposed for VoIP; H.323, SIP and MGCP. MGCP is perfect server-client protocol, so MGCP is very attractive VoIP protocol to ISP. This paper uses MGCP and offers modified MGCP for conference call. So that, Modified MGCP is compatible to MGCP, and supports conference call using IP-multicast.

• 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 Korea Society of Digital Industry and Information Management
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• v.7 no.2
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• pp.39-46
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• 2011

Privacy IP hacking problems such as invasion of privacy, password cracking, voice wiretapping and internet over charged occurred, because VoIP internet voice phone service gradually spread. This thesis attempted to attack the VoIP service network by application. First use application to spoof IP address then attempted wiretap the VoIP service and sends a lot of messages to disturb service movement. At this point, we connected VoIP soft terminal, so we can operate real-time filtering operator to block the SIP Flooding offence by monitor the traffic and detect the location where it got attacked. This thesis used experiment to prove it is possible to detect the offence and defend from SIP Flooding offence.

• Proceedings of the Korea Society of Information Technology Applications Conference
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• 2002.06a
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• pp.51-52
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• 2002

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.7
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• pp.10-18
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• 2013

With advancing of mobile communication services and commercializing of VoLTE (Voice of LTE), it is getting to pay attention on QoS of VoLTE. This paper proposes an integrated E-model in which some factors influenced to service quality of VoIP and VoLTE based voice communication system are considered in calculating the voice quality of Wideband Codec. The model aims to calculate R value which reflects the situations of access network, network characteristics, terminals' usage and mobility. We mainly deal with the integrated E-model's structure, related algorithms and optimal parameters for VoLTE. Some experiments show that the voice quality difference between VoIP and VoiceChecker, and VoLTE and POLQA, is below 10%. With the proposed model, we can calculate the voice quality by making use of the factors directly affected to service quality and the environment of VoLTE terminal and network. As a result, we can estimate the service quality in advance, without measuring it in real wireless environment.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.8
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• pp.1679-1684
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• 2004

This paper designs and implements a VoIP module for embedded system. Many other VoIP modules for desktop environment has widely implemented but the one for embedded system is not prevalent yet. The implemented VoIP module uses the OpenH.323 and is operated on the WinCE PDA. We use the TCP-friendly adaptation for congestion control. The experiment results show smooth communication but a little noise. Future works include the minimization of noise.