• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.2
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• pp.257-262
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• 2012

In this paper, transmission performance for serval VoIP traffics is meaured and analysised over MANET(Mobile Ad-hoc Networks) under some types of intrusions. VoIP simulator based on NS-2 is used for performance measure. In the simulation, MOS, network delay, packet loss rate and call connection rate is measured for transmission performance. With analysis of measued data, characteristics of each traffics for transmission performance parameters is showed on MANET under multi intrusions, As a results of this paper, some recommendation condtion for codec selection and blackhole intrusion is suggested to implement VoIP services on MANET under multi intrusions like as DDoS and blockhole intrusions.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.11
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• pp.2405-2410
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• 2012

The 4th generation mobile communication system, LTE, does not support an additional core network to provide voice service, and it is merged into a packet network based on all IP. Although Voice service over LTE can be supported by VoIP, it will be provided by the existing 3G networks because of the discontinuity of LTE coverage. However, it is inevitable to adopt VoIP over LTE to provide high quality voice service. In this paper, we investigate the capacity of VoIP over LTE. Our results indicate that spectral efficiency can be significantly improved as channel bandwidth increases in terms of VoLTE capacity. In addition, we can achieve higher VoLTE capacity without decreasing control channel capacity.

• ETRI Journal
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• v.38 no.5
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• pp.922-933
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• 2016

IP header compression schemes offer a valuable measure for bandwidth preservation. Such schemes have been practically implemented in infrastructure-based IP networks for point-to-point links. However, minimal research and practical implementation efforts have been conducted in the direction of an IP header compression strategy that can meet the peculiar requirements of multi-hop ad hoc wireless networks. In this paper, we present a practically implemented multi-hop IP header compression scheme using the Robust Header Compression (ROHC) protocol suite. The scheme runs on a novel identifier (ID) based networking architecture, known as an ID-based ad hoc network (IDHOCNET). IDHOCNET additionally solves a number of bottlenecks of pure IP-based ad hoc networks that have emerged owing to IP address auto-configuration service, distributed naming and name resolution, and the role of an IP address as an identifier at the application layer. The proposed scheme was tested on a multi-hop test bed. The results show that the implemented scheme has better gain and requires only O (1) ROHC contexts.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.4
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• pp.83-89
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• 2011

Nowadays, Internet Telephony Services based on SIP are gaining an explosive increase for general user. Thus a various demands of the user about convenience of IP-PBX are growing. The most important requirements of these is hybrid IP telephony system which is connected not only internet telephone but also general PSTN telephone. Therefore, in this paper, we have developed Hybrid IP-PBX connected all type of telephony terminals for small office. We have developed FXS module for connecting all type and FXO for using PSTN telephone number also. This Hybrid IP-PBX that is can be connected Soft-phone provide various optional services. We measured voice quality and test simultaneous calling for proof of validity.

• The KIPS Transactions:PartD
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• v.14D no.5
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• pp.565-572
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• 2007

In this paper, we developed a VoWLAN(Voice over WLAN) system based on an open source software. The system aims to provide VoIP service over wireless LAN with an IP-PBX server. The features of system presented in this paper are as follows. First, the initial cost for the development is reduced since the system is developed based on open source software. Second, the system provides various additional services such as Voice Mail, Conference Call, and Interactive Voice Response with a software IP-PBX server. Third, the VoWLAN terminal provides high-level user applications with minimal system resources using lightweight open software solutions. Finally, it is highly scalable since it is based on the open source software.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.8
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• pp.1415-1419
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• 2008

In this paper, we investigate various Audio Transmission Systems to implement Audio Transmission System based on AoIP of Internet transmission technology TCP/IP Network and we design and implement a Audio transmission system based on AoIP by adopting the most efficient one. The implemented system can be applied for various professional audio systems with large-scale audio distribution network as well as small-scale PA systems. Moreover, this can be applied in various fields which need audio transmission.

• 지역과문화
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• v.7 no.4
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• pp.69-81
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• 2020

As internet companies advance into the film industry, China's IP movies have grown rapidly and brought huge business benefits to the movie market. This paper tried to diagnose the phenomenon of the Chinese IP film craze and derive its meaning and implications. To this end, we selected representative cases to examine the problems and the corresponding improvement measures. Among the three genres of novels, cartoons, and music that are currently drawing the most attention in IP film adaptation, the examples were "Wu Kong", "Take My Brother Away", "My Old Classmate". The research results show that there are problems such as the unification of Chinese IP films, the lack of screenplay ability, and the copyright. Some improvement measures such as establishing IP copyright trading platform, strengthening IP creation, and promoting the value chain of IP industry are put to solve these problems. We hope that China's IP film market create a value chain on the basis of this paper.

• The Journal of the Acoustical Society of Korea
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• v.29 no.5
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• pp.347-353
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• 2010

In VoIP system, the speech quality is seriously degraded due to packet loss, and the degree of degradation by each packet loss depends on the characteristics of the corresponding packet. Therefore, it is possible to improve the speech quality of VoIP by selectively controlling the packet to be lost during transmission based on the expected degradation by the loss of each packet. In this paper, a new scheme to improve speech quality of DiffServ-based VoIP by assigning priority to each packet is proposed, and a method to determine the priority of each packet is developed. The performance of proposed method was measured in packet loss environment based on Gilbert model, and it was verified both objectively and subjectively that the speech quality is improved by the proposed method.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.6
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• pp.73-80
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• 2011

In this paper, we design and implementation of effective remote management system for VoIP(Voice over IP) terminal. Ordinary VoIP terminal is connected inside of NAT(Network Address Translation) assigned private IP address. NAT provides address mapping between outside public IP address and inside private IP address, for outside system telephone call is connected to inside VoIP terminal. In this paper, we also design and implementation of UDP hole punching function that the outside calls pass through the NAT, for outside management system call is connected to internal VoIP terminal, and gathered to management information of the VoIP terminal. Also, we evaluated and analysed of developed system in the test environment. As a result of test, It showed that it was well performed without any data error and data loss in the connectivity, and It showed that it was well gathered management information of the VoIP terminal.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.12
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• pp.167-173
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• 2015

This paper presents a study on the Full-HD HEVC(High Efficiency Video Coding) encoder IP(Intellectual Property) design. The designed IP is for HEVC main profile 4.1, and performs encoding with a speed of 60 fps of full high definition. Before hardware and software design, overall reference model was developed with C language, and we proposed a parallel processing architecture for low-power consumption. And also we coded firmware and driver programs relating IP. The platform for verification of developed IP was developed, and we verified function and performance for various pictures under several encoding conditions by implementing designed IP to FPGA board. Compared to HM-13.0, about 35% decrease in bit-rate under same PSNR was achieved, and about 25% decrease in power consumption under low-power mode was performed.