• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.5B
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• pp.430-441
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• 2006

As the various A/V devices and home networks are delivered to users, home networks are changing to an entertainment network. It is expected that the required network bandwidth and the amount of usage of media content in home entertainment networks will be increased. Although the access networks and home networks becoming a high speed network, there remains the problems for QoS-guaranteed media content transfer in home networks. Also, in the home network, there can be network traffic caused by applications like video conferencing, video telephone, and VoIP(voice over IP) as well as inner network traffic of home network. Since media content transfer requires the real-time delivery, it is very important and basic requirement that is to transfer media content to A/V device user wants while keeping the media quality. Even though there are many middleware protocol for home networking, they provide basic device discovery and control or simple functions for QoS-guaranteed media content transfer that are not enough to provide QoS-guaranteed media transfer service that user wants. Thus, in this paper, we propose the technique based on UPnP(universal plug and play) protocol for QoS-guaranteed media content transfer in the home network. The proposed technique is compatible with UPnP and can be used with UPnP as additional functions. In this paper, we utilize VideoLAN application to verify the proposed technique. We add the additional modules that support the proposed technique's function to VideoLAN and we verify the its functions through various test scenarios.

• Journal of the Korea Institute of Military Science and Technology
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• v.24 no.6
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• pp.645-654
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• 2021

In this paper, we proposed a protocol to convert 2W telephone analog signals to Ethernet data in a private PSTN 2W tactical voice system. There are several kinds of operational problems in the tactical telephone network where 2W telephone copper lines are installed hundreds of meters away from the PBX in a headquarter site. The reason is that it is difficult to install and maintain the 2W telephone copper cable in severe operational fields and to meet safety and stability operational requirements of the telephone line under lighting and electromagnetic environments. In order to solve these challenging demands, we proposed an efficient method that the 2W analog interface signals between a private PBX system and a 2W telephone is converted to Ethernet messages using the optical Ethernet data communication network already deployed in the tactical weapon system. Thus, it is not necessary to install an additional optic cable for the ethernet telephone line and to maintain the private PSTN 2W telephone network. Also it provides safe and secure telecommunication operation under lightning and electromagnetic environments. This paper presents the conversion protocol from 2W telephone signals over Ethernet interface between PBX systems and 2W telephones, the mutual exchange protocol of ethernet messages between two converters, and the rule to process analog signal interface. Finally, we demonstrate that the proposed technique can provide a feasible solution in the tactical weapon system by analyzing its performance and experimental results such as the bandwidth of 2W telephone ethernet network and the transmission latency of voice signal, and the stability of optic ethernet voice network along with the ethernet data network.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.5 s.347
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• pp.77-83
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• 2006

In this paper, we designed, constructed and evaluated the system for the FTTH suited to a residence and apartment using CWDM-PON techniques. These systems have capacity to service at 100Mbps rate to 384 users in the same breath. Also, the services include the internet, CATV, IPTV and wireless LAN. In the case of ire network, the data could be transmitted by UTP cable and optical fiber and case of wireless one, the data transmitted using WLAN. The distance between the cental office and the user is 20km and the data rate is 100Mbps maximum. Of course, the optical network used just one fiber optical core. For the basic material, we obtained the characteristics of optical transceiver module, Mux/Demux and transmission qualities depends on the environment.

• Journal of Advanced Marine Engineering and Technology
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• v.35 no.8
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• pp.1098-1104
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• 2011

Data communications networks at sea can be modelled by multi-layered networks with traditional carriers (RF, satellites), and BWA (wireless LAN, WiBro, LTE), which partially makes it possible the high speed communication services (WWW, VoIP) at sea. In this paper, a novel routing scheme (CPR) is proposed which selects an optimal carrier for each hop in routes based on carrier preferences (CP). The carrier preferences are measured proactively depending on the feasibility of transmission characteristics (transmission rate, cost, and latency time) of the carriers for each application. Performance was compared with that of the OMH-MW (Optimal Medium per Hop based on Max-Win) routing scheme.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.9 no.6
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• pp.599-603
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• 2016

Recently the wireless LAN system is substituting wired LAN system notably as the number of mobile users increases greatly along the advancement of technology. But the wireless LAN has a critical weakness in the security such as data leakage. Thus a safe security system is imperative to avoid threatening from hackers with offering the best convenience to inner users. In this research, we have developed a RADIUS wireless LAN security system for industrial applications, which performs the EAP authentication with the compatibility for any maker of wireless LAN. The system has interfaces based on WEB, providing DB access function for user management so that users can perform authentication of 802.1x in their computers.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.9 no.4
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• pp.929-935
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• 2008

As internet telephony services based on Session initiation Protocol (SIP) enter the spotlight as marketable technology, many products based on SIPs have been developed and utilized for home and office telephony services. The call connection of an internet phone is classified into specific call connections and group call connections. Group call connections have a forking function which delivers the message to all of the group members. This function requires excessive message control for a call connection and creates heavy traffic in the network. In the internet cail system model. most of the call-setup messages are directed to the proxy server during a short time period. This heavy message load brings an unwanted delay in message processing and. as a result, call setup can not be made. To solve the delay problem, we simplified the analysis of the call-setup message in the proxy server, and processed the forking function distributed for the group call-setup message. In this thesis, a new system model to minimize the load is proposed and the subsequent implementation of this model demonstrates the performance improvement.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.9A
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• pp.663-668
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• 2003

With the advent of high speed networking technology, computers connected to the high-speed networks tend to consume more of their CPU cycles to process data. So one of the solutions to improve the performance of the computers is to reduce the CPU cycles for processing the data. As the consumption of the CPU cycles is increased in proportion to the number of the packets per second to be processed, reducing the number of the packets per second by increasing the length of the packet is one of the solutions. In order to meet this requirement, two types of jumbo packets such as jumbograms and jumbo frames have already been standardized or being discussed. In case that the jumbograms and general packets are interleaved and scheduled together in a router, the jumbogrms may deteriorate the QoS of the general packets due to the transfer delay. They also frequently exhaust the memory with storing the huge length of the packets. This produces the congestion state easily in the router that results in the loss of the packets. In this paper, we analyze the problems in processing the jumbo packets and suggest a noble solution to overcome the problems.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2A
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• pp.128-139
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• 2004

As the Internet usage grows, people want to access the Internet while they are moving. To satisfy this requirement economically, IEEE 802.11 Wireless LANs(WLANs) are rapidly deployed. In order to support mobility, WLANs must provide smooth handoff mechanism. Recent studies show, however, handoff delay of WLANs exceeds 300ms, most of which is due to slow scanning mechanism finding a new AP. With this handoff delay, current WLANs is not suitable to provide seamless realtime interactive services such as VoIP sevice. In this paper, we analyze the current handoff method of IEEE 802.11 and we propose a new handoff algorithm which can decrease time needed for searching a new AP and thus reduce overall handoff time. We show by simulation that the proposed algorithm has shorter handoff delay than current handoff method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.8B
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• pp.507-513
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• 2005

As the rapid increasement of the number of internet users has occured recently, many multimedia application services have been emerging. To improve quality of service, traffic can be suggested to be classified with priority in EPON(Ethernet Passive Optical Network), which is supporting the multimedia application services. In this paper, multimedia application services treat bandwidth classifying device in serving both delay sensitive traffic for real-time audio, video and voice data such as Von(Voice over Internet Protocol), and for real-time traffic such as BE(Best Effort). With looking through existing mechanisms, new mechanism to improve the quality will be suggested. The delay performances and packet losses of traffic achieved by supporting bandwidth allocation of upstream traffic in suggested mechanism will be analized with simulations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.7B
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• pp.528-535
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• 2012

This paper proposes silence drop first(SDF) active buffer management algorithm to increase the voice capacity when silence suppression is used. This algorithm finds and drops silence packet rather than voice packet in the queue for resolving buffer overflow of queue. Simulations with voice codec of G.729A and G.711 are performed. By using proposed SDF algorithm, the voice capacity is increased by 84.21% with G.729A and 38.46% with G.711. Further more, SDF algorithm reduces the required link capacity and loosens the silence packet inter-arrival time limit to provide target voice quality compared with that of conventional algorithms.