• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.637-639
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• 2002

현재 인터넷은 저렴한 사용료와 정보에 대한 욕구를 충족시키는 방대한 공개 자료를 바탕으로 그 규모를 더욱 확대하여 가고 있으며, 이러한 인터넷 사용자의 확대는 새로운 서비스에 대한 요구를 창출하게 되었다. 이러한 저렴하고 규모가 큰 인터넷을 이용하여 기존의 통신망을 대체하는 연구가 활발히 이루어지고 있으며, VoIP(Voice over Internet Protocol)가 인터넷의 대표적인 서비스로 등장하고 있다. VoIP 서비스에 대한 연구는 IETF와 ITU가 중심이 되어 이루어지고 있으며 IETF에서 제안한 MGCP, SIP 와 ITU에서 제안한 H.323 과 같은 프로토콜을 기반으로 VoIP 서비스를 위한 다각적인 접근과 연구가 진행중이다. 본 연구는 VoIP 서비스를 위한 여러 프로토콜 중 IETF가 주관하고 있는 MGCP(Media Gateway Control Protocol ) 스팩에 따라 MGCP를 이용한 ADSL-VoIP Gateway를 개발하여 보다 효율적인 망 자원 활용을 가능하게 하며, 향후 제공될 다양한 음성／동영상 서비스에 대한 기반을 마련하고자 한다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.430-433
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• 2001

Voice of IP(VoIP) technology Provides voice service as well as data service via Internet. It has been a promising technology as Internet grows fast and the requirements are increasing. Among the standardized protocols for VoIP, H.323 has been one of the most developed technologies and widely used. Recently, the requirements for standardization of inter-domain interoperability is raised, since it has been developed by focusing on intra-domain service. To meet such requirements, IMTC(International Multimedia Telecommunications Consortium. Inc.) and VoIP Forum have elaborated on interoperability standard based on H.323, respectively in international and domestic markets. In this paper, we describe the call model and specify required functionalities of inter-domain interoperability based on H.323.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.2
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• pp.279-287
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• 2001

In this paper, we discuss the scale of subjective speech quality measurement over VoIP(Voice over IP) network which is a component of broadband networks. Objective parameters of multimedia services like PSNR or jitter can easily measured and defined, but these factors are not easily meet the user's perceptual recognition. We suggest the speech quality measurement scale through the subjective measurement for end-to-end speech quality composed of sender-side quality, transmission quality, receiver-side quality, which provide the degree of correctness of representation of speaker, the degree of impairment caused by various factors, the degree of recognition of processed speech, respectively. Also, we examined the proposed method and verify it's availability.

• Journal of Information Technology Services
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• v.1 no.1
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• pp.57-66
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• 2002

We are looking at the two different kinds of IT on this paper. One is VoIP and the other is VR (voice recognition). We are more interesting at the evolving techniques and services produced by combining the two techniques mentioned above. Recently, there are so many services and products appeared in the market using voice recognition technique. Now the technique has progressed on the level that can even replace the user interfaces using the QUI or general DTMF. Therefore, we are expecting so many various new services showed UP In the market which is combination of the VoIP and VR. Up until now, three models are available in the field which are wired telephone, wireless telephone, and wireless internet. We know the effectiveness of the VoIP is maximized more when this technique is combined with others rather than used alone without other techniques.

• Proceedings of the Korean Information Science Society Conference
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• 2007.06d
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• pp.229-233
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• 2007

VoIP(Voice over Internet Protocol)란 음성 데이터를 IP 데이터그램 방식으로 기존 인터넷망을 통해 전달해 주는 기술을 말한다. 기존 인터넷망을 이용하여 음성 데이터를 전달해 줌으로써 기존의 음성 전화 서비스에서 사용되던 회선비용을 크게 절감할 수 있다는 점은 VoIP의 장점 중 하나이다. 그런데 VoIP를 기존의 인터넷망에 그대로 적용하기에는 VoIP에서 사용되는 프로토콜의 특성으로 인해 어려움이 따르게 된다. 즉, 기존의 인터넷망에서 사용되고 있는 방화벽과 NAT(Network Address Translator)장비는 보안을 위해서는 필수적인 요소들 이지만, VoIP의 통신 입장에서는 음성 데이터의 원활한 통신을 방해하는 요소로 작용을 하게 된다. 이러한 문제는 VoIP 통신에 사용되는 시그널링 프로토콜인 H.323과 SIP 프로토콜의 연결 설정과 데이터 전송에 사용되는 동작 방식이 방화벽과 NAT장비의 기능에 충돌하는 점 때문에 발생하게 된다. 따라서 기존의 인터넷망을 그대로 사용하면서 VoIP의 통신이 원활하게 이루어지도록 하기 위해서는 이러한 문제의 해결이 반드시 이루어져야 한다. 본 논문에서는 기존에 Firewall/NAT Traversal 문제 해결을 위해 연구되던 기법들에 대해 살펴보고, 새로운 구조를 제시한다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.8
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• pp.1339-1346
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• 2006

This thesis outlines a preparation plan for e-Trade business service regarding tendency development in super-highway information network including internet, cable, and wireless communication. It also explains two perspectives in e-Trade market: changes of circumstances and consumerism, and revitalization devices of Von(Voice over Internet Protocol) service technology for creating new market in rapidly changing IP(Internet Protocol) environment. Plus it illustrates what core competence and progress business organizations must have in current situation, forecasts turns of future e-Trade market, and analyzes technological applications of VoIP service in an extended viewpoint of corporate strategy.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.2
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• pp.243-248
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• 2012

In this paper we simulate the capacity of VOIP calls through WiFi network by computer simulations using OPNET modeler. The results show that sudden quality degradations occur on all VoIP calls when the number of call of an AP(Access Point) increases beyond a specific value. The reason of the quality degradation was turned out to be the queueing delay at the down link of AP. Under the IEEE 802.11b environments, the maximum number of VoIP calls of an AP maintaining the required voice quality (MOS > 2.5), was evaluated as 5, 12, and 27 when we use G.711, G.729a, and G.729a VAD codec, respectively.

• Journal of information and communication convergence engineering
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• v.7 no.3
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• pp.309-313
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• 2009

In this paper, end-to-end VoIP(Voice over Internet Protocol) performance is evaluated by simulation with NS-2 simulation tool. There are many results studied and published for VoIP performance over TCP/IP networks. But, almost all of them were focused on wired or wireless Internet environments. About MANET (Mobile Ad Hoc Network), VoIP is currently studying several points of research. In this paper, analysis of VoIP performance is done with focusing on the mobility of MANETs. MOS(Mean Opinion Score), network delay, packet loss rates are considered as end-to-end QoS performance parameters.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.5
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• pp.129-132
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• 2009

Frame erasure is one of the most difficult problems in voice over IP (VoIP) networks and is a major source of speech quality degradation. In this paper, a frame erasure concealment algorithm based on nonlinear regression analysis is presented to minimize speech quality deterioration in code-excited linear prediction (CELP) based coders. We applied the proposed scheme to the ITU-T G.729 standard and obtained improved perceptual evaluation of speech quality (PESQ) scores compared to the conventional methods.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.6
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• pp.89-94
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• 2015

FMS which has a concept that wireless terminal can replace wire terminal services is a technologies that is can provide service costs same as wire terminal in the special zone. Enterprise type of FMC that is developed making up for the weak point is must have to improve voice quality and FMC performance in the soft phone. This paper measure voice quality based on the one way of the total estimated delay time of FMC to carry out IMS services between IP-PBX and FMC soft-phone to operate it's controller optimally and put forward evidence to be in 120ms and 150ms in the VoIP FMC voice quality. To measure FMC performances in four categories evaluated trials and prove its performances.