• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.10a
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• pp.459-461
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• 2010

Ad-Hoc VoIP system is an important IP phone technique used in MANET networks. In small-scale networks, VoIP system could be built easily while keeping the Qos on a acceptable level. Although Ad-Hoc VoIP system is designed as 'Client-Server' model, due to the property of MANETs that no Base Stations working as servers, we have the SIP Server with promary funtions worked on mobile terminals instead. In this paper, we explored how to build VoIP systems in MANETs and achieved the construction from within linux.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.777-779
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• 2009

Study for VoIP in MANET is mainly focused on transmission performance or voice quality of VoIP traffic as single traffic. In this study, VoIP performance is analyzed of MANET which has several types of traffics. Diffserv scheme is used to process the traffics. It is showed that VoIP traffic performance is improved by differv scheme, as a result of analysis for VoIP and FTP traffic. Simulator, which is implemented with NS-2 package and VoIP patch, is used.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.71-77
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• 2011

In this paper, we propose a conversational quality measurement (CQM) system for providing the objective QoS of high quality mobile VoIP voice telecommunication. For measuring the conversational quality, the VoIP telecommunication system is implemented in two smart phones connected with VoIP. The VoIP telecommunication system consists of echo cancellation, noise reduction, speech encoding/decoding, packet generation with RTP (Real-Time Protocol), jitter buffer control and POS (Play-out Schedule) with LC (loss Concealment). The CQM system is connected to a microphone and a speaker of each smart phone. The voice signal of each speaker is recorded and used to measure CE (Conversational Efficiency), CS (Conversational Symmetry), PESQ (Perceptual Evaluation of Speech Quality) and CE-CS-PESQ correlation. We prove the CQM system by measuring CE, CS and PESQ under various SNR, delay and loss due to IP network environment.

• International journal of advanced smart convergence
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• v.1 no.2
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• pp.59-63
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• 2012

The mVoIP service is the technology which focuses on the wireless data service as the IP network having the transmission rate of 100 Mbps classes on the high-speed middle of movement through the WiFi, WiBro and 3G mobile radio communication network, and etc. and is developed. Since 2010, the mVoIP (mobile VoIP) service shows the rapid growth due to 4G-LTE service seriously disclosed from July with the Smart phone and 2,011 it begins to be rapidly popularized. In this research, additionally the mVoIP service industry activation plan is presented with the trends of technology development including the chip-set/module/terminal, etc. based upon local and foreign market trend searchlight through the market demand analysis. The mVoIP service downloads App to the mobile apparatus and or can provide the service as the software through the WiFi network. Therefore, the change which is large in the products development aspect is to be have no. Expected to is being provided as the added service in the case of 4G-LTE as a matter of course, the service deployment where it is based on the market principle and demand needs is needed.

• The KIPS Transactions:PartC
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• v.17C no.3
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• pp.299-306
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• 2010

IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

• 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.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.4 s.346
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• pp.1-12
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• 2006

In this paper we propose an efficient mobile multimedia communication protocol, mobile terminal software platform and mobile VoIP application for IP-based convergence networks. The Proposed mobile multimedia communication protocol is called as ST-MRSVP (Split tunnel based Mobile Resource reServation Protocol) which integrates split tunnel based Mobile IP and RSVP in order to support hish speed mobility. Since mobile terminal platform supports QoS (Qualify of Service) with keeping seamless mobility, mobile QoS supporting modules are developed and interworked together by means of shared memory mechanism. Testbed is composed of a core-network embedding the proposed protocols and wireless LAN-based access networks. We verify functionality and performance of the proposed techniques by using various mobility test over the testbed. As a result, the proposed architecture can reduce the handover delay time with QoS support under 30% comparing with the standard mechanisms and support voice quality as good as CDMA phone.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.10a
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• pp.153-155
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• 2010

Internet phone (VoIP) services in Korea have achieved noticeable growth year after year since the service launching, and the growth still coninues. The market of mobile internet phone also expands sharply. Therefore, it is crucial to deploy networks which can support mobile internet phone services with excellent quality. For mobile internet phone services, it will be necessary to build and use networks with good mobility and high transmission rate. Current wireless networks for Internet services include 3G, Wi-Fi, and mobile WiMAX networks. 3G provides good mobility but lower transmission rate, whereas Wi-Fi exhibits excellent transmission rate but less mobility. Mobile WiMAX networks taking the merits of both, high mobility and transmission rate, are being deployed widely in recent years. This article examines the recent standardization efforts of WiMAX Forum for VoIP service in WiMAX networks.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.11b
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• pp.807-810
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• 2003

현재 무선랜은 AP(Acess Point)간 이동시 2계층 로밍을 이용하여 이동성을 제공한다. 그러나 무선랜의 AP간 로밍은 같은 서브넷 내에서는 동일한 IP로 MAC Layer의 접속만을 변경하여 이동성을 제공할 수 있지만, 다른 서브넷으로 이동할 경우에는 DHCP를 이용하여 IP를 새로 할당 받아야 한다. 최근 Hotspot 등 무선랜이 활성화 되어감에 따라 서브넷간 이동시에도 IP 이동성 제공을 위해서는 무선랜에 Mobile IP 도입이 필요하다. 또한 무선 인터넷 사용자의 급증으로 차세대 인터넷 프로토콜로 주목받고 있는 IPv6를 이용하여 이동성을 제공하려는 연구가 활발히 진행되고 있다. 본 논문에서는 SIP 기반의 VoIP 단말기에 FMIPv6를 적용하여 VoIP의 서브넷 간이동성을 제공하기 위하여 핵심프로토콜인 FMIPv6 프로토콜과 SIP 프로토콜을 설계하였다.

• Journal of Information Technology Applications and Management
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• v.14 no.4
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• pp.221-233
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• 2007

Which factors should influence consumer consideration to subscribe to Voice over Internet Protocol (VoIP)? Policy issues, managerial concerns, and demographic variables are possible factors. This paper discusses policy and managerial issues regarding VoIP adoption. A model that explains VoIP adoption is proposed and tested. This study analyzes a survey of 750 prospective VoIP users in Korea. The testing is accompanied by logistic regression and discriminant analysis. The results show that trust in VoIP, relative comparison of Quality to fixed service, numbering plan, satisfactions of call Quality and customer services on both fixed and mobile services have impacts on the adoption of VoIP. Implications for VoIP providers and policy makers are presented.