• Annual Conference of KIPS
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• 2002.11a
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• pp.53-56
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• 2002

VoIP(Voice over IP) 기술은 다양한 인터넷 응용 서비스 보급의 대중화에 기여한 주요 통신기술의 하나이다. 이에 따라 인터넷 이용자가 급격히 증가하고 인터넷 전화의 수요가 증가하게 되었다. 특히, 음성 뿐만 아니라 영상에 대한 기술이 접목되어 V2oIP(Voice and Video over IP) 라 일컬어 지는 기술이 보급되면서, 인터넷 영상전화에 대한 대중화가 이루어 지고 있다. 한편, 다양한 인터넷 응용 서비스 보급의 대중화에 따른 가정 내에서의 네트워크 환경 또한 부각되고 있어 정보가전분야에 큰 변화가 일고 있다. 이에 홈서버 중심의 홈네트워크환경이 구축된다. 따라서, 기존의 pc 를 단말기로 한 인터넷 환경 및 제반 응용 서비스가 그대로 홈서버 중심의 홈네트워크환경으로 옮겨져야 할 필요가 있다. 본 논문에서의 전체 시스템은 임베디드 리눅스 기반인 Qplus 운영체제를 기반으로 하는 홈서버상에 HoCoS(Home Collaboration Service) 시스템이 탑재되며, 이 시스템은 크게 영상전화 시스템과 공동작업 시스템으로 구성된다. 본 논문에서는 상기 시스템 중 SIP(Session Initiation Protocol)기반의 홈서버용 영상전화 시스템에 대한 설계 및 구현에 대해 기술하고자 한다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.2
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• pp.254-259
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• 2008

Mesh WLANs, which consist of wireless mesh routers connecting each other in a mesh topology and self-operate after their autoconfiguration, have several advantages in convenience, swiftness and flexibility of deployment and operation over existing WLANs the expansions of which are done by connecting the APs with wires. However, many technical issues still remain to be solved. Among them, network performance degradations due to the interference between the adjacent hops in multi-hop mesh WLANs, and the reusability of the existing wireless network protocols are critical problems to be answered. This work evaluates the VoIP support performance of IEEE 802.11a/g-based mesh WLANs with multiple wireless interfaces with simulations. The results show that there exit an unfairness in VoIP packet delay performances among mobile routers located at different hops, and that although the capacity of the admitted calls can be increased by increasing the size of voice packet payload it is far less than the expected one. This suggests that the existing 802.11 MAC protocols have their limitation when applied in mesh networks and their enhancement or even a newer one nay be required.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.261-264
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• 2011

Corded telephone to the phone using a wireless phone as the trend to switch, free Wi-Fi-enabled mobile phones, netbooks, and mobile devices, are spreading rapidly. But wireless Internet phone calls using your existing Internet network to deliver Internet services because it has a vulnerability that will occur. Government agencies are using Voice over Internet Protocol(VoIP) calls from the current wired and wireless connection and usage is increasing. In this paper, we have discovered that the vulnerability of wireless internet Wi-Fi AP and the FMC administrative agencies, such as VoIP on your wireless device to study the vulnerability. Wi-Fi AP and the FMC is to analyze the vulnerability. VoIP call interception, attack, attack on the base of the experiment is the analysis. Security-enhanced VoIP call for a Wi-Fi AP and the FMC's defense against man-in-the-middle attacks and is the study of security measures.

• The Journal of the Korea Contents Association
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• v.7 no.3
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• pp.34-41
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• 2007

The presence service which provides user's presence information by subscription and notification is one of SIP(session initiation protocol) extension services, and it is used importantly in VoIP(Voice over IP) and Instant Messaging service. In this paper, we propose a new method in which users can combine and control presence service and call processing services in various ways by extending call processing language, and only changed parts of the presence information are published instead of full presence information document. Each user registers full presence information document with his own call processing script during the first publication to a presence server. The presence server executes these call processing scripts, so it can provide various services with combination of presence service and call processing services during the presence subscriptions and notifications. Afterwards, users publish only changed parts of the presence information and the presence server notify only these changed parts to watchers. Therefore the efficiency of the overall system can be improved. The performance of our proposed model is evaluated by experiments.

• The KIPS Transactions:PartD
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• v.16D no.6
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• pp.945-960
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• 2009

The purpose of this study is to define determinants of VoIP user acceptance and to verify significant causality among latent variables - performance expectancy, effort expectancy, cost expectancy, social influence, facilitating conditions, behavioral intend, use behavior - based on UTAUT model. We presented the expanded hypotheses including the new factor, cost expectancy and analyzed the moderating effect of user age, gender and usage experience variables. For a accuracy of predicted results, we focused on survey analysis with 641 real user samples. Compared to previous studies, it is meaningful that this research verified the conceptual difference between behavioral intention and usage behavior. As a result, all proposed hypotheses accepted and moderating effects are supported significantly in age and use experience moderating variables.

• Journal of Korean Institute of Industrial Engineers
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• v.32 no.1
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• pp.1-8
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• 2006

A modified PCF(Point Coordination Function) protocol with the optimal polling sequence is defined in detail and shown to improve the efficiency of the conventional PCF protocol in IEEE 802.11 wireless LAN standard. The problem for the optimal polling sequence is formulated as TSP(Travelling Salesman Problem) with the distance values of 1's or 0's. Numerical examples show that the optimal polling sequence increases the capacity of the real-time service such as VoIP(Voice over Internet Protocol).

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.127-134
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• 2005

Lost packet robustness is an most important quality measure for voice over IP networks(VoIP). Recovery of the lost packet from the received information is crucial to realize this robustness. So, this paper proposes the lost packet recovery method from the received information for real-time communication for CELP coder. The proposed BS-PLC (Both Side Packet Loss Concealment) based WSOLA(Waveform Shift OverLab Add) allow the lost packet to be recovered from both the 'previous' and 'next' good packet as the LP parameter and the excitation signal are respectively recovered. The burst of packet loss is modeled by Gilbert model. The proposed scheme is applied to G.729 most used in VoIP and is evaluated through the SNR(signal to noise) and the MOS(Mean Opinion Score) test. As a simulation result, The proposed scheme provide 0.3 higher in Mean Opinion Score and 2 dB higher in terms of SNR than an error concealment procedure in the decoder of G.729 at $20\%$ average packet loss rate.

• Annual Conference of KIPS
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• 2006.05a
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• pp.639-642
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• 2006

본 논문에서는 다양한 서비스가 운용되고 있는 인터넷 망에서 PCM 및 ADPCM으로 압축된 음성데이터를 전송할 경우에 발생하는 패킷 크기와 한계 지연시간의 변화가 수신측의 음질에 미치는 영향을 분석하였다. 이를 기반으로 주어진 한계 지연시간에 대하여 적절한 음질을 제공하기 위한 전송패킷의 크기에 대하여 분석하여 적절한 코덱 선택 방법을 제안하였다. 제안된 방법의 실험결과를 입증하기 위해 음질 평가인 MOS평가 방법을 사용했으며 측정방법으로는 서울을 중심으로 전국5개 지역 지점별 5회 측정 각 지점의 임의 번호를 서울 콜 센터로 Call Forwarding 설정 후 측정하고, VQT은 PAMS 알고리즘과 ADRA(Audio Direction Reference Audio)를 사용하여 측정한 결과 음성코덱의 데이터비와 Datagram size에 의해 음성 품질이 달라짐과 적절한 코덱 선택방법임을 확인하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.176-179
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• 2003

The use of Multiparty Conference service based on SIP for VoIP provides is gradually magnified, and the work for continuous development and standardization on SIP is in the process of advancing. But, currently it is impossible for SIP to support identity discovery and distribution of each participant for multiparty conference. In this paper, we propose a SIP extension for guaranteeing security on the multiparty conference using SIP by adding new method and reconstructing header informations. With this, it is also possible to identify discovery and to distribute each participant using SIP extension when a call is established for closed multiparty conference.

• The Journal of the Korea Contents Association
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• v.1 no.1
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• pp.130-136
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• 2001

Real-time applications like VoIP in mobile networks need smooth handovers in order to minimize or eliminate packet loss as a mobile host(MH) transitions between network links. In this paper, we design a new variable buffering mechanism for IPv6 by which an MH can request that the router on its current subnet buffers pad(eta on its behalf while the MH completes registration procedures with the router of a new subnet. Performance results show that our proposed queueing scheme with a variable space allocation is quite appropriate for mobile internet environment in terms of the packet loss rate.