• Review of KIISC
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• v.16 no.1
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• pp.59-63
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• 2006

광대역통합망(Broadband Convergence Network. BcN)의 핵심서비스로 자리 잡고 있는 VoIP(Voice over Internet Protocol)는 IP망 경유를 통한 저렴한 통신비용 및 다양한 부가서비스 제공가능성에 따라 서비스 사업자뿐만 아니라 기업에서도 점차 도입이 증가할 것으로 예상되는 서비스이다. 그러나, 보안솔루션의 인터넷 전화 트래픽 통과 문제 및 IP망에서 발생 가능한 공격뿐만 아니라 VoIP 스팸이나 도청 같은 새로운 보안이슈들이 많이 발생할 것으로 예상됨에 따라 본 논문에서는 VoIP 서비스에 대한 기밀성, 무결성, 가용성적인 측면에서 예측 가능한 취약성을 살펴봄으로서 더 좋은 서비스를 제공하는데 이바지할 수 있을 것이다.

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.7
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• pp.773-778
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• 2014

IDS(Intrusion Detection System) can be used as a countermeasure for blackhole attacks which cause degrade of transmission performance by causing of malicious intrusion to routing function of networks. In this paper, effects of IDS for transmission performance based on mobility patterns is analyzed for MANET(Mobile Ad-hoc Networks), a suggestion for effective countermeasure is considered. Computer simulation based on NS-2 is used in performance analysis, VoIP(Voice over Internet Protocol) as an application service is chosen for performance measure. MOS(Mean Opinion Score), call connection ratio and end-to-end delay is used as performance parameter.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.2A
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• pp.12-19
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• 2005

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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4A
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• pp.310-315
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• 2002

The Bark Coherence Function (BCF) defies a coherence function within perceptual domain as a new cognition module, robust to linear distortions due to the analog interface of digital mobile system. Our previous experiments have shown the superiority of BCF over current measures. In this paper, a new BCF suitable for VoIP is developed. The unproved BCF is based on the wavelet series expansion that provides good frequency resolution while keeping good time locality. The proposed Wavelet based Bark Coherence function (WBCF) is robust to variable delay often observed in packet-based telephony such as Voice over Internet Protocol (VoIP). We also show that the refinement of time synchronization after signal decomposition can improve the performance of the WBCF. The regression analysis was performed with VoIP speech data. The correlation coefficients and the standard error of estimates computed using the WBCF showed noticeable improvement over the Perceptual Speech Quality Measure (PSQM) that is recommended by ITU-T.

• Proceedings of the Korea Multimedia Society Conference
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• 2000.11a
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• pp.299-302
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• 2000

인터넷에서 음성 서비스를 제공하는 인터넷 텔레포니 또는 VoIP(Voice over IP) 기술은 대부분 ITU-T H.323을 기반으로 제공되고 있다. 그러나 H.323은 그 구조가 복잡하기 때문에 이해하는데 상당한 노력과 오랜 개발 기간이 요구된다. IETF는 이러한 문제를 극복하고 인터넷 환경에서 잘 동작할 수 있는 IP 텔레포니용 프로토콜로서 Session Initiation Protocol (SIP)을 표준화하였다. 본 논문에서는 VoIP 서비스를 위한 SIP의 구현 사항을 기술한다

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.839-843
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• 2002

In this paper, we present a next generation VoIP network architecture. Specifically, we present key components such as SoftSwitch and media gateway, and important protocols for VoIP services. We also present basic components and mechanisms for VoIP QoS. The architecture presented in this paper is able to support open interfaces and multimedia traffic, and therefore various multi-services can be supported efficiently using the architecture.

• Journal of Digital Contents Society
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• v.12 no.1
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• pp.85-94
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• 2011

QoS(Quality of Service) is defined as "The collective effect of service performance which determines the degree of satisfaction of a user of the service" by ITU-T Rec. E.800. While the use of VoIP(Voice Over Internet Protocol) has been widely implemented, persistent problems with QoS are a very important sue which needs to be solved. This research is finding the assignment of VoIP QoS to deduct how to manage the control system and presenting the QoS control process and framework under NGN(Next Generation Network) environment. The trial framework is the modeling of the QoS measurement metrics, instrument, equipment, method of measurement, the series of cycle & the methodology about analysis of the result of measurement. This research underlines that the vulnerability of the VoIP protocol in relation to its QoS can be guaranteed when the product quality and management are controlled and measured systematically. Especially it's very important time to maintain the research about VoIP QoS measurement and control because the big conversion of new network technology paradigm is now spreading. In addition, when the proposed method is applied, it can reduce an overall delay and can contribute to improved service quality, in relation to signal, voice processing, filtering more effectively.

• Journal of the Korea Computer Industry Society
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• v.5 no.9
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• pp.1017-1024
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• 2004

In this paper, we propose the progressed interworking method between H.323 and SlP, then explain the improved property. The VolP(Voice over Internet Protocol) technology which is able to use a voice service through internet is more cheaper then existing telephone charges, and is easil)· accept the various of multimedia services from internet. Previous connectionmethod of VoIP used H.323 protocol, but it is very complex to connection establishment. so, the SIP(Session Initiation Protocol) protocol that propose in SIP-Working Group is in use recently. Therefore, we need new interworking methodology between H.323 and SIP Products. In this thesis, the progress interworking method between H.323 and SIP are Propose, then interpret unnecessary packet delay for call setup and improved feature of message exchange.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2018.10a
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• pp.568-570
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• 2018

Currently, with rapid progress and supply of mobile communication technology, LTE(Long Term Evolution) technology is expanded and widely used to industrial and emergency communications beyond earlier smart-phone based service. In this paper, transmission performance of voice traffic, one of railway communication service based on LTE-R as an application field of LTE technology, is analyzed. This study is performed performance analysis with level of application service and consider effects of satisfaction level for users. Computer Simulation based on ns(Network Simulation)-3 is used for analysis and VoIP(Voice over Internet Protocol) specification is used for voice traffics. Results of this paper is used to implement LTE-R networks and develope application services over LTE-R network.