• Journal of Korea Multimedia Society
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• v.10 no.3
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• pp.335-347
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• 2007

VoIP (Voice over IP) is a technology to transport video and voice traffic over IP networks such as Internet. Today, the VoIP technology is viewed as the right choice for providing voice, video, and data communication among various terminals over the next generation network. This paper discusses a multipoint VoIP implementation with end-point mixing model which can support multipoint conference without a conference bridge. The multipoint VoIP is implemented with SIP (Session Initiation Protocol), and supports STUN (Simple Traversal of UDP Through NATs) since it works in an asymmetric NAT (Network Address Translator) environment. The characteristics of this paper are as follows. It is possible that all terminals in the hierarchical conference don't receive the duplicated media information because we use the end-point mixing model with the new media processing module. And, the paper solves the problem that the hierarchical conference session should be separated into several sessions when a mixing terminal terminates the hierarchical conference session.

• Journal of Internet Computing and Services
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• v.5 no.6
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• pp.31-43
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• 2004

Voice over Internet Protocol (VoIP) is currently a popular research topic as a real time voice packet transmission method. But current Internet environment do not guarantee the quality of voice when we take a side view of delay, jitter and loss. Up to now, many voice based evaluation algorithms have been used to measure speech quality of VoIP systems. However, these algorithms have the defects that their results are different according to voice samples and some algorithms can not take network environment for speech transmission path. The E-model can be used to solve the problems of these algorithms. In this paper. we introduce VoIP planning guidelines through the various analysis of E-model which can model impairments of network quality as well as VoIP equipment quality systematically, We, also, show the evaluation method and results of speech transmission quality.

• The Journal of the Acoustical Society of Korea
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• v.37 no.6
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• pp.512-517
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• 2018

Improving voice quality is a major concern in telecommunications. In this paper, we propose a robust speech quality enhancement against background noise and packet loss at VoIP (Voice-over-IP) receiver. The proposed method combines network jitter estimation based on hybrid Markov chain, adaptive playout scheduling using the estimated jitter, and speech enhancement based on restoration of amplitude and phase to enhance the quality of the speech signal arriving at the VoIP receiver over IP network. The experimental results show that the proposed method removes the background noise added to the speech signal before encoding at the sender side and provides the enhanced speech quality in an unstable network environment.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.6
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• pp.1327-1332
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• 2015

In this paper, we studied several aspects that affect the quality of VoIP using WiFi network. It's clear that the background data traffic within an AP, the end-to-end delay and the traffic loss of TCP/IP network gives serious effects on the voice quality. A kind of access control for the VoIP connection within an AP should be done for the acceptable voice quality.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.05a
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• pp.414-418
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• 2003

VoIP is a core technology used to transmit both voice and data in an integrated packet' form. Within this technology, SIP is the signaling protocol used for 'real time' call services; particularly those where H323 is used. Yet, when considering the needs of mobile users, it is essential we integrate VoIp within the mobile technology so the mobile host is able to receive the 'packet' transported and by, and connected to, any available internet-address. For all this to occur, we need to improve Network Delay by reducing transmission problems associated with mobile services. If we are to obtain an optimal service then we must reduce any network delays which may arise from joining Mobile IP and VoIp services. This paper, therefore, considers how, unlike previous research, these delays may be improved through the use of the signaling technology\ulcorner SIP. It also considers how this research may be introduced into current wired and wireless integrated services enabling them to use the IP 'packet'.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.5-8
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• 2000

The voice over IP(VoIP) technology is currently recognized as the base technology for the next generation telecommunication services. So the VoIP market has been extremely expanding with the opportunity for cheap phone calls. This paper describes the packet voice testing issues and scenarios for the VoIP services. These issues and scenarios are deduced from the testing results through KT VoIP testbed composed of commercial systems.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.5
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• pp.1109-1116
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• 2010

In this paper, some performance characteristics of VoIP(Voice over Internet Protocol) for MANET(Mobile Ad-hoc Networks) with simulation is studied and appropriate condition for implementation of VoIP service is suggested. VoIP simulator is implemented with NS(Network Simulator)-2. VoIP traffic for simulation is generated with some codecs of G.711, G.723.1, G.726-32, G.729A, GSM.AMR and iLBC. As simulation results for traffic transmission under $670{\times}670m$ 50node MANET environment, performance data for MOS(Mean Opinion Score), network delay, packet loss rate and transmission bandwidth are measured. Normalized analysis about measured results shows that maximum VoIP connection satisfying VoIP service quality condition is 15.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.862-865
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• 2002

VoIP was simply used to transfer voice call between IP host. VoIP provides efficiently end-to-end voice communication, however it needs to supply value-added services to users. This raper describes design of SIP communication for registering CPL.

• Journal of Advanced Navigation Technology
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• v.12 no.1
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• pp.28-33
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• 2008

VoIP provided voice service using Internet is receiving footlights when it escapes an initial curiosity. VoIP interest became larger, because it can transfer existing phone service and deliver voice data through internet technology. Is inheriting as it is threats of IP base because a VoIP technology provides audiocommunication service taking advantage of an existing IP technology, and have new threats that happen from new know-hows for VoIP service offer. In this paper, presented about problem and consideration and countermeasure of examines about security threat of VoIP service, and applies security.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.4
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• pp.385-390
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• 2010

VoIP Service is provided voice & image call using internetwork unlike traditional call. VoIP usage is becoming generalization & development. As a result, threats of security are steadily increasing. Regarding this situation, we will investigate the security problem of VoIP in various aspects. Also We will investigate main technology of security solution method for solve this problem.