• 한국음향학회지
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• 제37권6호
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• pp.512-517
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• 2018

음성 품질의 향상은 통신 분야의 주요 관심사이다. 본 논문에서는 VoIP(Voice-over-IP) 수신부에서의 배경잡음 및 패킷손실에 강인한 음질향상 방식을 제안한다. 제안된 방식에서는 하이브리드 마르코프 체인 기반 네트워크 지터추정, 추정된 지터를 이용한 적응적 플레이아웃 스케줄링, 그리고 진폭 및 위상 복원 기반의 음성 향상 방식 등을 결합하여 IP 네트워크를 통해 VoIP 수신부에 도착하는 음성신호의 품질을 향상시킨다. 실험결과는 제안된 방식이 송신부의 인코딩 전에 음성신호에 추가된 잡음을 제거하고 불안정한 네트워크 환경에서 양질의 음성을 제공하는 것을 확인할 수 있다.

• Journal of information and communication convergence engineering
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• 제7권3호
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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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• 제1권1호
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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.

• 한국정보처리학회:학술대회논문집
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• 한국정보처리학회 2011년도 춘계학술발표대회
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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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• 제16권2호
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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.

• 한국융합학회논문지
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• 제10권8호
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• pp.9-14
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• 2019

현재, 5G 상용화 성공에 따라 IoT 기술이 고도화 되고 있어 다양한 사물인터넷 공공 안전 서비스를 통합한 서버 시스템이 개발되어야 한다. 본 논문에서는 IP PBX를 기반으로 IoT 디바이스와 게이트웨이를 연결하는 IoT 플랫폼인 공공 안전 통합서버를 구현하였다. 본 서버는 임베디드 OS 기반으로 다양한 IoT 서비스가 하나의 시스템에서 수행되고 공공장소의 비상통화 및 방송을 처리하는 호 처리/방송서버 기능을 내장하고 있고, IoT센서 데이터와 비상벨 정보를 수집하여 응급 상황 시 사고현장에서 비상알람과 대피방송 등을 자동송출 하며, 일상적인 정보는 상위의 IoT 서비스 서버에 전달하여 IoT서비스 서버의 명령에 따라 공공안전관리 서비스를 제공하여 국가 사회의 안전망 서비스를 편리하고 저렴하게 제공할 수 있다.

• 한국정보통신설비학회:학술대회논문집
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• 한국정보통신설비학회 2009년도 정보통신설비 학술대회
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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.

• 융합신호처리학회논문지
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• 제13권4호
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• pp.211-219
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• 2012

본 논문에서는 VLBI상관서브시스템(VLBI Correlation Subsystem, VCS)의 상관블록 TCP/IP 출력데이터의 무결성 검사를 위한 소프트웨어의 개발과 상관출력 데이터의 손실을 방지하기 위한 성능개선 방법에 대해 기술한다. VCS의 상관결과는 TCP/IP 패킷 통신으로 데이터아카이브(Data Archive)에 저장된다. 본 논문에서는 데이터아카이브에 저장된 상관결과의 무결성을 확인하기 위해 VCS의 TCP/IP 패킷 정보를 이용한 무결성 검사 소프트웨어를 개발하였다. 개발한 소프트웨어를 이용하여 3단계의 무결성 검사 과정을 제안하고, 상관처리 실험을 통하여 제안방법의 유효성을 확인하였다. 또한 VCS와 데이터아카이브 사이에는 최소 적분시간 이내에 TCP/IP 패킷 통신이 완료되어야 하지만, 짧은 적분시간에 다량의 패킷과 대용량 데이터로 인해 패킷 손실이 발생할 뿐만 아니라 상관결과의 무결성 문제에도 영향을 미치는 것으로 확인되었다. 본 논문에서는 TCP/IP 패킷 손실의 원인을 분석하고 VCS의 FPGA(Field Programmable Gate Array) 설계에 대한 수정방법을 제시하여 상관결과의 무결성 문제를 해결하고자 한다.

• Journal of information and communication convergence engineering
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• 제6권1호
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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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• 제9권4호
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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.