• Journal of KIISE:Computing Practices and Letters
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• v.14 no.3
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• pp.261-265
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• 2008

As increasing the user of VoIP service using ITSP(Internet Telephony Service Provider), the VoIP spam becomes a big problem. The spam used in the existing public telephone is detected by using the pattern inspection of call behavior because it is difficult to filter contents for the characteristic of real-time voice communication. However there is a false-positive problem. The threat on spam remains where spam with low threshold can't be detected or users share one number. In this paper, we propose anti-spam for VoIP based on luring test. The proposed method gives a user luring test and he/she can connect to a receiver if passing turing test. A ticket is given to a user that pass luring test and it reduces overhead of luring test in re-dial. The proposed method is implemented on ASUS WL-500G wireless router and Asterisk IP-PBX. Experimental results show the effectiveness of the proposed method.

• Annual Conference of KIPS
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• 2007.05a
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• pp.1021-1024
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• 2007

VoIP 서비스는 인터넷을 기반으로 하므로 인터넷망에서 발생하는 보안 위험이 내재해 있고, 서비스가 실시간으로 이루어진다는 특성으로 인해 기존의 보안 솔루션으로 이러한 보안 위험을 해소하기는 어렵다. 따라서 VoIP 망 구축 초기단계부터 정보보호를 고려하여 보안대책을 세우고 이를 반영하는 것이 비용과 보안 효과 측면에서 바람직하다. 본 논문에서는 상용 VPN 제품에 공개 프로젝트인 SIP VoIP Gateway 'Asterisk'을 연동하여 사용자 인증과 데이터 기밀성을 효과적으로 수행하기 위한 VoVPN(Voice over Virtual Private Network)을 설계하고 구현하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2005.11a
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• pp.898-900
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• 2005

초고속 통신망의 확대 적용으로 인터넷의 빠른 성장과 함께 음성과 비디오 그리고 데이터를 통합하고자 하는 노력이 시도 되고 있다. VoIP(Voice over IP)는 IP를 이용하여 음성과 데이터를 패킷 형태로 통합하여 실시간으로 전송하는 기술이다[1]. 패킷 네트워크에서 VoIP 시그널링 기술을 이용하면 망 자원의 효율적 이용 및 PSTN에 가까운 음질 그리고 인터넷과 연계한 다양한 음성서비스 지원(H.323, SIP, MGCP 등 다양한 신호처리 지원)이 가능하다. 본 논문에서는 VoIP망에서 IP기반 녹취 시스템을 설계 및 구현한다. 녹취 시스템은 고객과 상담원의 통화 내용을 자동으로 녹음하여 보관함으로써 고객의 요구사항을 명확히 파악할 수 있으며 녹취 데이터의 통계 자료 제공으로 효율적인 관리가 지원되고 선택 녹취, 스케줄링 녹취, 상담원의 평가 자료를 제공하여 고객 관리의 질적인 향상을 지원한다. 본 논문의 녹취 시스템은 고객과의 통화 내용을 녹취하여 서버의 녹취 DB에 저장하여 관리하는 녹취 시스템으로 모든 네트워크 환경에서 사용할 수 있으며 CTI와 연동하여 효율적이고 체계적인 녹취 시스템 구국이 가능하다.

• Journal of Korea Multimedia Society
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• v.7 no.7
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• pp.896-904
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• 2004

In this paper, We present an efficient variable rate speech coder for spectral efficiency and packet processing technique for packet loss compensation of a voice codec with 10msec frame in VoIP service. Through disconnecting the users from the spectral resource during silence interval of about 60% period, a variable rate voice coder based on a voice activity detection(VAD) can increase spectral gain by two times. The performance of the method was analyzed by variation of detected voice activity factor and degraded speech frame ratio under various background noise level, and compared those of G.729B of ITU-T 8kbps standard speech codec. A method to compensate lost packets utilized addition of recovery data to a main stream and error concealment scheme for speech quality enhancement, the performance is verified by reconstructed speech quality. The proposed scheme can achieve spectral gain by two times or enhance speech quality by 3dB through reserved bandwidth of VAD. Therefore, the proposed method can enhance a spectral efficiency or speech quality of VoIP.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.12
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• pp.1187-1192
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• 2016

Performance analysis results are limited to of network level, because network level transmission parameters are used for performance measure and analysis of network design, construction and operation on underwater MANET, With this way of performance analysis based on network level, it is not easy to analyze transmission performance related with user level transmission quality. In this paper, transmission performance focused on application traffic be required by user is investigated to supplement weakness of performance analysis based on network level. Voice traffic, which is expected to be increasingly used on underwater MANET, is considered as application service, Some conditions for underwater MANET will be proposed to support transmission quality, MOS, CCR and EED, etc.. A computer simulation based on NS-2 is used for performance measure, voice traffic is generated as VoIP specification.

• Review of KIISC
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• v.17 no.5
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• pp.57-65
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• 2007

초고속 인터넷의 보급 확산과 IT기술의 급격한 발전으로 우리 사회에서 인터넷 이용이 보편화를 넘어 필수적인 요소로 자리 잡고 있다. 이러한 현상에 따른 이용자 증가로 인해 최근 들어 패킷 망에 음성을 실어 보내는 VoIP(Voice Over Internet Protocol) 기술이 주목을 받고 있다. 이 기술로 인해 저렴한 통신비용 및 다양한 부가 서비스의 제공 가능성에 따라 새로운 비즈니스 모델이 증가할 것으로 예상되고 있다. 그러나 VoIP 서비스는 기존 인터넷망에서 발생할 수 있는 보안 취약성뿐만 아니라 인터넷 전화 트래픽 통과 문제 및 VoIP스팸이나 도청 같은 기존에 없었던 새로운 형태의 보안 이슈들이 많이 발생할 것으로 예상한다. 본 논문에서는 VoIP 신규 보안 위협을 분석하고, 분석된 보안 위협을 바탕으로 VoIP 공격 패킷 발생 도구를 구현하여 실제 공격 시 기존 보안 장비 시스템의 대응 여부에 대해서 기술하고자 한다.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.3 no.3
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• pp.251-265
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• 2009

VoIP service is the transmission of voice data using SIP protocol on an IP-based network. The SIP protocol has many advantages, such as providing IP-based voice communication and multimedia service with low communication cost. Therefore, the SIP protocol disseminated quickly. However, SIP protocol exposes new forms of vulnerabilities to malicious attacks, such as message flooding attack. It also incurs threats from many existing vulnerabilities as occurs for IP-based protocol. In this paper, we propose a new virtual proxy to cooperate with the existing Proxy Server to provide state monitoring and detect SIP message flooding attack with IP/MAC authentication. Based on a proposed virtual proxy, the proposed system enhances SIP attack detection performance with minimal latency of SIP packet transmission.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.14 no.7
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• pp.3238-3246
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• 2013

The progress of Information Technology enables people can communicate each others using the Internet. As a smartphone proliferates, many free mobile VoIP(Voice over IP) application are developed and accepted by smartphone users. In this study we investigate the factors affecting the acceptance of mobile VoIP applications and the structural relationship among these factors. We review the related works and extract the related factors and build a research model describing the causal relationship among these factors. We conduct an empirical study - a survey and statistical analysis - to verify the research model. EFA(Exploratory Factor Analysis) is applied for variables in the survey and SEM(Structural Equation Model) is used to reveal the structural relationship among the factors. We can find that two factors - usefulness of mobile VoIP and social influence - positively affect usage intention and actual use. These findings imply that it is required to emphasize the benefits of mobile VoIP use and add S/W functionalities enhancing social influence.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.637-640
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• 2011

Recently, rapid spreading of smart-phone make an expectation for increase of practical application for MANET(Mobile Ad-Hoc Network) which is not used infrastructure like as base-station. One of important application for MANET will be VoIP(Voice over Internet Protocol) known as Internet telephony. On the other hand, information intrusions causing serious problems is not allowed exception of MANET. Especially, there are some dangerous problems of intrusions to MANET, differently to other networks, because of it's usage of military purpose or emergency application of rescue. In this paper, to analyze this intrusion problem on MANET with blackhohe attacks, effect of intrusion to transmission performance is studied. VoIP traffic is assumed as an application service on MANET, computer simulation with NS-2 is used for this analysis.