• 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.

• Journal of information and communication convergence engineering
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• v.1 no.1
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• pp.6-11
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• 2003

In general, standard TCP/IP (transmission control protocol-internet protocol), which is called TCP/IP, is using as the communication basis protocol between any collections of networks that is connected. In this paper, using this TCP/IP implementation of remote control system and suitable program for long distance communication is proposed. This system can make system, which basic Ethernet and TCP/IP used system, to mini modeling, so all module that is using here can be used. Therefore, intention of this paper is to reduce expenses, to effective manage for plant and to increase of productivity as linking each plant of several factory to TCP/IP and Ethernet, and then many control plant and manager minimize the needed course.

• Proceedings of the Korea Information Processing Society Conference
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• 2011.04a
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• pp.905-908
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• 2011

와이파이 기능이 탑재된 모바일 기기 보급이 확산되면서 무선네트워크를 이용한 많은 서비스가 개발되고 있다. 그중 기존 전화망(PSTN)에서 발전하여 인터넷 네트워크를 이용한, 음성과 데이터 네트워크 융합의 대표적인 인터넷 전화(VoIP)서비스 이용률이 증가하고 있는 추세다. VoIP 기술은 FMC(Fixed Mobile Convergence) 서비스의 기반이 되며, 이에 따라 FMC서비스는 기존의 VoIP 보안위협 및 특성을 상속 받게 된다. 본 논문은 유무선 통합에 의한 여러 가지 유무선 단말, 네트워크 및 서비스 특성에 대한 보안 위협을 상속 받게 되는 FMC 환경에서의 VoIP보안 위협을 소개하고 보안 요구사항을 설계한다. 또한 안전한 FMC서비스를 위해 총체적인 보안망 설계 시 VoIP보안 위협 및 보안요구사항에 적합한 보안솔루션의 아키텍처 구조를 제안한다.

• International Journal of Internet, Broadcasting and Communication
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• v.16 no.2
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• pp.10-21
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• 2024

Recently, audio systems are changing the configuration of conventional sound reinforcement (SR) systems and public address (PA) systems by using audio over IP (AoIP), a technology that can transmit and receive audio signals based on internet protocol (IP). With the advancement of IP technology, AoIP technologies are leading the audio market and various technologies are being released. In particular, audio networks and control hierarchy over peer-to-peer (Anchor) technology based on AoIP is a system that transmits and receives audio signals over a wide bandwidth without an audio mixer, creating a novel paradigm for existing audio system configurations. Anchor technology forms an audio system by connecting audio sources and output equipment with On-site audio center (OAC), a device that can transmit and receive IP. Anchor's receiving OAC is capable of receiving and mixing audio signals transmitted from different IPs, making it possible to configure a novel audio system by replacing the conventional audio mixer. However, Anchor technology does not have the ability to provide audio effects to input devices such as microphones and instruments in the audio system configuration. Due to this, when individual control of each audio source is required, there is a problem of not being able to control the input signal, and it is impossible to individually affect a specific input signal. In this paper, we implemented a tone control module that can individually control the tone of the audio source of the input device using the audio processor core in the audio system based on Anchor technology, tone control for audio sources is possible through a tone control module connected to the transmitting OAC. As a result of the study, we confirmed that OAC receives the signal from the audio source, adjusts the tone and outputs it on the tone control module. Based on this, it was possible to solve problems that occurred in Anchor technology through transmitting OAC and tone control modules. In the future, we hope that the audio system configuration using Anchor technology will become established as the standard for audio equipment.

• Journal of the Korea Convergence Society
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• v.10 no.8
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• pp.9-14
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• 2019

Currently, with the success of 5G commercialization, a server system that integrates various Internet public safety services should be developed. In this paper, we developed a public safety integrated server, which is an IoT platform connecting IoT device and IoT gateway based on IP PBX. This server is based on embedded OS and various IoT services are executed in one system and call processing / broadcasting server function that processes emergency call and emergency broadcasting in public places is built in. This system collects IoT sensor data and emergency bell information and automatically sends out emergency alarms, emergency evacuation broadcasts, etc. at an accident site in an emergency situation, and transmits the daily information to the upper IoT service server, Provide public safety management services.

• 한국정보통신설비학회:학술대회논문집
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• 2009.08a
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• pp.68-71
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• 2009

All-IP network requires change of the existing IP network engineering methods as the convergence service market between communication and broadcasting industries using IP network is growing rapidly. Especially the video services like IPTV require more strict transmission quality and higher bandwidth than the existing data services. So it is difficult to design All-IP network by the over-provisioning method which used to be used for the existing IP network design. It also requires a heavy investment which becomes one of big obstacles to the IPTV service expansion. In order to reduce the investment costs, it is required to design an optimized network by maximizing the utilization of the network resources and at the same time maintaining the customer satisfaction in terms of service quality. In this paper, we first analyze the effects of IPTV traffic on the existing internet. Then we compare two traffic engineering technologies, which are dimensioning without admission control and dimensioning with admission control, on the All-IP network design by simulation. Finally, we suggest cost effectiveness of traffic engineering technologies for designing the All-IP network.

• Journal of the Institute of Convergence Signal Processing
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• v.13 no.4
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• pp.211-219
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• 2012

In this paper, we described the software development for verifying the integrity of output data of TCP/IP for VLBI Correlation Subsystem (VCS) correlation block and proposed the performance improvement method in order to prevent the data loss of correlation output. The VCS correlation results are saved at the Data Archive system through TCP/IP packet transmission. In this paper, the integrity verification software is developed so as to confirm the integrity of correlation result saved at the data archive system using TCP/IP packet information of VCS. The 3-step integrity verification process is proposed by using the developed software, its effectiveness was confirmed in consequence of correlation experiments. In addition, TCP/IP packet transmission must be completed within minimum integration period. However, there is not only TCP/IP packet loss occurred but also the problem of correlation result integrity affected in account of a large quantity of packets and data during short integration time. In this paper, the reason of TCP/IP packet loss is analyzed and the modified methods for FPGA(Field Programmable Gate Array) of VCS are proposed, the integrity problem of correlation results will be solved.

• Journal of information and communication convergence engineering
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• v.6 no.1
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• pp.68-72
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• 2008

The VoIP(Voice over IP) has been worldwide used and already put to practical use in many fields. However, it is needed to ensure secret of VoIP call in a special situation. It is relatively difficult to eaves-drop the commonly used PSTN in that it is connected with 1:1 circuit. However, it is difficult to ensure the secret of call on Internet because many users can connect to the Internet at the same time. Therefore, this paper suggests a new model of Internet telephone for eavesdrop prevention enabling VoIP(using SIP protocol) to use the VPN protocol and establish the probability of practical use comparing it with Internet telephone.

• International journal of advanced smart convergence
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• v.9 no.4
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• pp.209-214
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• 2020

VoIP technology has been widely used for exchanging voice or image data through IP networks. VoIP technology, often called Internet Telephony, sends and receives voice data over the RTP protocol during the session. However, there is an exposition risk in the voice data in VoIP using the RTP protocol, where the RTP protocol does not have a specification for encryption of the original data. We implement programs that can extract meaningful information from the user's dialogue. The meaningful information means the information that the program user wants to obtain. In order to do that, our implementation has two parts. One is the client part, which inputs the keyword of the information that the user wants to obtain, and the other is the server part, which sniffs and performs the speech recognition process. We use the Google Speech API from Google Cloud, which uses machine learning in the speech recognition process. Finally, we discuss the usability and the limitations of the implementation with the example.

• Proceedings of the Korea Multimedia Society Conference
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• 2012.05a
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• pp.353-354
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• 2012

광대역 VoIP 네트워크 환경에서는 불안정한 네트워크 환경으로 인해 음성패킷이 불규칙적으로 수신되어 음성데이터의 재생이 원활하지 못하다. 이러한 문제점을 해결하기 위해 본 논문에서는 네트워크 상태에 따라 원활하게 음성패킷을 재생시키는 스케줄링 방식을 제안한다. 제안하는 방식은 수신단에 도착한 패킷 헤더정보를 이용해 네트워크 지터를 추정하고, 추정된 지터와 지터버퍼와 음성프레임버퍼에 존재하는 패킷수 및 음성프레임 개수, 음성클래스정보에 따라 음성프레임의 길이를 변화시켜 재생시킴으로써 수신단의 버퍼링 지연을 줄이고 출력신호의 음성왜곡을 최소화한다. 제안하는 스케줄링 방식의 성능측정을 위해 버퍼링 지연과 PESQ를 기존 음성패킷 재생 스케줄링 방식과 비교한다.