• Journal of the Korea Convergence Society
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• v.8 no.12
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• pp.93-99
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• 2017

The elevator system combines various convergence technologies with the development of ICT technology. Emergency call devices which are safety related devices is applied as an obligation of the elevator and those scope also varies. In this paper, we propose an integrated model that overcomes the limitations of existing voice emergency call devices and efficiently manages and manages video call based service structures in VoIP based on wired and wireless environments. This method effectively manages and operates various lift data and video records in the elevator between the manager, the server and the user. And also It is possible to secure the quality of video call in VoIP and cloud service environment and increase the reliability of safety management and enhance various service environment by creating an integrated structure utilizing various data and additional services in the elevator.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.12C
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• pp.1238-1244
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• 2002

In this paper, H.235-based security mechanism for H.323 multimedia applications was implemented. H.235 covers authentication using HMAC, Diffie-Hellman key exchange, session key management for voice channel, and encryption functions such as DES, 3DES, RC2. Extra encryption algorithms such as SEED, and AES were also included for possible use in the future. And, we also analyzed the quality of service (QoS), the requirement of implementation, and interoperability to the result in this study. The results could be applied to secure simple IP phone terminals, gateways, or gatekeepers.

• Management & Information Systems Review
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• v.28 no.3
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• pp.191-210
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• 2009

Internet was first designed only for sending data, but as the time passed, internet started to evolve into a broadband multi-media web that is capable of transmitting sound, video, high-capacity data and more due to the demands of internet users and the rapid changing internet-communication technology. Domestically, in January, 2000 Saerom C&T, launched a free VoIP, but due to limited ways of conversation(PC to PC) and absence of a revenue model, and bad speech quality, it had hit it's growth limit. This research studied VoIP based on technological enhancement in super-speed internet. According to IDC, domestic internet market's size was 80,800 million in 2008, and it formed a percentage of 12.5% out of the whole sound-communication market. in case of VoIP, it is able to maximize it's profit by connecting cable and wireless network, also it has a chance of becoming firm-concentrated monopoly market by fusing with IPTV. Considering the fact that our country is insignificant in MVNO revitalization, regulating organizations will play a significant roll on regulating profit between large and small businesses. Further research should be done to give VoIP a secure footing to prosper and become popularized.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.5
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• pp.1131-1136
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• 2015

VoIP and IP Telephony telephony technology development is a growing by easy to using IP-PBX by using phone from using existing lines rather than the internet. IP-PBX do not use the phone line from phone work for many companies and institutions of management costs reduce as provides similar to regular phone line quality. But IP-PBX to introduce for need to be the initial cost on is should buy for expensive hardware equipment or commercial software. In this paper, suggest way to introduce IP-PBX do not buy expensive hardware equipment or commercial software. Suggest IP-PBX on designed and implement for IP-PBX server using Raspberry Pi and Asterisk. And verification treatise on the suitability of conducted by voice calls based on IP-PBX between PC and a Smartphone

• Journal of the Korea Safety Management & Science
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• v.11 no.3
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• pp.199-207
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• 2009

IP Telephony service was restricted to an outgoing call and low quality since the trust domestic IP Telephony service launch of Saerome co. Ltd, in Jan. 2000. However, Interest of IP Telephony service, which is substituted for PSTN, has been highly elated because of the developed equipment softswitch and new technology. This kind of importance and marketing of VoIP are recognized to telecommunication providers. With this trend, they try to administrate customer satisfaction and invest R&D to survive in this hard competition and unexpected change. To achieve this objective, they should try to realize the searching process of the quality decision attribution (QDA). However, there is little research on the aspect of service quality of Internet telephony so far. For this, the investigator established the tangibles, the reliability, the responsibility, the assurance, the empathy, the charge with information sources as core elements. In order to examine the influence of IP Telephony service upon the attitudes toward a brand and the purchase intention.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.8 no.5
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• pp.1-12
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• 2008

The multimedia streaming service in NGN architecture is not only to deliver video streaming (VoD, Broadcasting TV, etc.) but also to provide new services and service bundles, such as Triple Play (IPTV + VoIP + Internet). Among these services, Internet Protocol Television (IPTV) is becoming a convergence of communication, content, computing, as well as an integration of broadcasting and telecommunication services. In this paper, we addresses enhanced IPTV management scheme aspect of E2E from Home Network to Head End Center over NGN to support efficient service management with a full quality of service (QoS) guarantee.

• Information Systems Review
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• v.10 no.3
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• pp.39-64
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• 2008

As digital convergence evolves, it is expected that the market of IP-based services like VoIP and IPTV will be expanded. In particular, IPTV market is expected to attract consumers' attention through various interactive services offering a variety of experiences to consumers. Interactivity sets apart old media from new one in terms of how to mediate effects of user satisfaction. The object of this study is to investigate (1) multi-dimensional Interactivities in an interactive medium based on IP and relationship among them, and (2) significant factors affecting cognitive absorption of interactive media users. This study aims to provide implications on how to develop strategies for IP-based media including e-learning system.

• Journal of Communications and Networks
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• v.9 no.4
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• pp.446-456
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• 2007

Increased access to broadband networks has led to a fast-growing demand for voice and video over IP(VVoIP) applications such as Internet telephony(VoIP), videoconferencing, and IP television(IPTV). For pro-active troubleshooting of VVoIP performance bottlenecks that manifest to end-users as performance impairments such as video frame freezing and voice dropouts, network operators cannot rely on actual end-users to report their subjective quality of experience(QoE). Hence, automated and objective techniques that provide real-time or online VVoIP QoE estimates are vital. Objective techniques developed to-date estimate VVoIP QoE by performing frame-to-frame peak-signal-to-noise ratio(PSNR) comparisons of the original video sequence and the reconstructed video sequence obtained from the sender-side and receiver-side, respectively. Since processing such video sequences is time consuming and computationally intensive, existing objective techniques cannot provide online VVoIP QoE. In this paper, we present a novel framework that can provide online estimates of VVoIP QoE on network paths without end-user involvement and without requiring any video sequences. The framework features the "GAP-model", which is an offline model of QoE expressed as a function of measurable network factors such as bandwidth, delay, jitter, and loss. Using the GAP-model, our online framework can produce VVoIP QoE estimates in terms of "Good", "Acceptable", or "Poor"(GAP) grades of perceptual quality solely from the online measured network conditions.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.349-360
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• 2002

In ITU H.323 teleconference system, the RTP/RTCP protocol is offered to transfer real-time multimedia stream. Both sender and receiver hate experience in packet loss and jitter which result from network congestion over Internet. Audio quality over Internet depends on the number of lost packets and on jitter between successive packets. The goal of our study is to improve the speech quality over Internet by checking the packet loss characteristics of the network and adopting the but for control management mechanism at the receiver. We suggest a dynamic redundant audio transmission mechanism which examines the packet loss rate and uses the feedback information through RTCP.

• The Journal of the Korea institute of electronic communication sciences
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• v.6 no.6
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• pp.823-829
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• 2011

This paper covers the service quality of packet-based voice service that is provided over a wireless broadband (WiBro) network. Using a measurement software that has been developed in the course of preparing a advanced service quality management scheme for the packet-based voice service over a wireless network[2][3], a huge scale of experiment is conducted to measure the real quality of the voice service. Based on our analysis of the measurement result, the service quality of the voice service is supposed to be quite good over WiBro networks. In addition, another experiment to investigate the effect of degradation of wireless transmission conditions on the service quality of the voice service shows the values of wireless service metris in which mean opinion score (MOS) starts to decrease.