• International journal of advanced smart convergence
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• v.11 no.2
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• pp.108-117
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• 2022

The recent development of information and communication technology has a great impact on the audio/video industry. In particular, IP-based AoIP transmission technology and AVB technology are making changes in the audio/video market. Video signal transmission technology has been introduced to the market through a network, but it has not replaced the video switcher function. Video signals in the conference room or classroom are still controlled by the switching device. In order to switch input/output video devices, a cable that is not limited by distance must be connected to the switcher. In addition, the control of the switching device must be performed by a person who has received professional training. In this paper, it is a technology that can be operated even by non-experts by replacing complex video cables (RGB, DVI, HDMI, DP) with LAN cables and enabling IP-based video switching and transmission (Video Mirroring over IP: VMoIP) to replace video switcher equipment. We are going to do this study, I/O videos were controlled in the form of matrix and high-definition videos were transmitted without distortion, and VMoIP is expected to become the standard for video switching systems in the future.

• International journal of advanced smart convergence
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• v.12 no.3
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• pp.1-12
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• 2023

A public address (PA) system installed in a building is a system that delivers alerts, announcements, instructions, etc. in an emergency or disaster situation. As for the products used in PA systems, with the development of information and communication technology, PA products with various functions have been introduced to the market. PA systems recently launched in the market may be connected through a single network to enable efficient management and operation, or use voice recognition technology to deliver quick information in case of an emergency. In addition, a system capable of locating a user inside a building using a location-based service and guiding or responding to a safe area in the event of an emergency is being launched on the market. However, the new PA systems currently on the market add some functions to the existing PA system configuration to make system operation more convenient, but they do not change the complex PA system configuration to reduce facility costs, maintenance, and management costs. In this paper, we propose a novel PA system configuration for buildings using audio networks and control hierarchy over peer-to-peer (Anchor) technology based on audio over IP (AoIP), which simplifies the complex PA system configuration and enables convenient operation and management. As a result of the study, through the emergency signal processing algorithm, fire broadcasting was made possible according to the detection of the existence of a fire signal in the Anchor system. In addition, the control device of the PA system was replaced with software to reduce the equipment installation cost, and the PA system configuration was simplified. In the future, it is expected that the PA system using Anchor technology will become the standard for PA facilities.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.1
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• pp.134-144
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• 2009

The Ubiquitous/Unified Communication Platform supports various multimedia communication tools such as VoIP, Email, Unified Messaging, Instant Messaging, Web Conferencing, Audio/Video Communication etc. In this paper we introduced the main functions and architecture of the Unified Communication Platform and we researched on the function analysis and design of the VoIP System that supports PC-to-PBX/PSTN Phone and PBX/PSTN Phone-to-PC communications through the connectivity and interoperation with PSTN.

• Journal of Digital Convergence
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• v.9 no.5
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• pp.225-233
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• 2011

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. Recently, the introduction of smart phones has led to a growth in social networking services and thus, the research and development of Internet telephony has been actively progressed and has the potential to become a replacement for the telephone service that is currently being used. In this paper we designed and implemented a recording system which records voice data of SIP-based Internet telephone's voice calls. It is developed on the linux system and has some features such as audio mixing of two in/out voice channels, live packet sniffing, and the ability to transfer mixed audio files to the log file server. These functions are implemented using various open source softwares. Afterwards, this VoIP recording system will be applied as a base technology to advanced services like a VoIP-based call center system.

• Proceedings of the Korea Information Processing Society Conference
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• 2013.11a
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• pp.257-258
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• 2013

VoIP(Voice over Internet Protocol)는 인터넷과 같은 IP 망에서 음성과 영상을 전송하기 위한 기술이며, 차세대 망에서의 음성, 테이터 통신을 위한 기술로 부상되고 있다. 따라서 VoIP 응용은 인터넷 망을 이용하는 다양한 단말기들 사이의 음성 및 영상 통신을 위하여 더욱더 많이 사용되어 질 것으로 예상된다. 본 논문에서는 VoIP 기술을 여러 분야에 적용할 수 있는 응용성과 실제 다자간 음성통신의 구현 방법에 대해서 기술하겠다.

• 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.

• The Journal of Society for e-Business Studies
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• v.9 no.4
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• pp.137-155
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• 2004

A video-conference system is being utilized in web based application services in various fields due to the widespread use of Internet and the progress of computer technologies. This system should use the public IP address for sharing file and white board and it is difficult to manage the internal network users of the firewall and non-public IP address users. In this paper, we propose an Application Level Gateway which transforms non-public IP address into public IP address. This mechanism is for the internal network users of the firewall or non-public IP address users over the Internet. We also propose a Control Daemon which manages video and audio media dynamically according to network bandwidth. This mechanism can start and terminate a video conference and manage the process of the video conference.

• The KIPS Transactions:PartC
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• v.17C no.3
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• pp.299-306
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• 2010

IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

• Journal of the Korea Society of Computer and Information
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• v.8 no.4
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• pp.117-125
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• 2003

The proposed mechanism has an group management server, which manages the information about both the receivers and the senders. Any receiver can dynamically join/leave the multicast VC. The signaling overload due to group membership changes is not concentrated on the sender, but it is distributed to many receivers for the scalability improvement. The associated signaling messages propagates from the receivers to the ATM switch dedicated to the multicast VC, and hence no signaling overload exists in the shared links there is no latency for the receiver to wait. Our proposed scheme is superior in the view of scalability, the efficiency and the latency to other schemes.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.4
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• pp.39-44
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• 2014

IMS(IP Multi-Media Subsystem) base on the IP service platform which can offer multimedia as the voice, audio, video, and data is service platform. In 3G mobile communication in the early day, IMS had a suggestion for supporting to multimedia service in the 3GPP. But now It is broadly substituting in the IPTV, wire phone company and it is substituted in internet platform base on the soft-switch in currently. Especially nowadays, 4G LTE in a mobile communication company is rapidly growing in market. Therefore, in this study, we had designed to the main prosser that can admit to 1,000 user over and SIP gateway which can link the IMS 코어 that can link SIP Device which adopt the standard protocol on the SIP.