• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.76-78
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• 2002

Multicasting over mobile IP network becomes more important with the increasing needs of supporting multimedia services in mobile network. The IETF has suggested two approaches which are remote subscription and bidirectional tunneling for supporting mobility management in multicasting over mobile IP. But these protocols have problems - the frequent reconstruction of multicast delivery tree, packet less during handoff, convergence problem, and so on. In this paper, we propose to use preconfiguration of multicast delivery tree when mobile host enters the foreign network. It will decrease the frequency of multicast delivery tree reconstruction, and reduce the packet loss during handoff, Also the multicast delivery tree maintained by Keep Alive messages makes the signaling overload of networks diminished.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.872-875
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• 2003

오늘날 인터넷기술의 급속한 발전으로 인하여 IP Phone서비스의 활성화가 기대되며 이러한 IP Phone 서비스시장의 확대는 기존 PSTN망에 의한 전화서비스시장에 지대한 영향을 미친다. 따라서 IP Phone 서비스시장에 있어서도 상호접속제도 도입의 필요성이 점증되고 있다. 이에 따라 본고에서는 IP Phone 서비스의 상호접속유형을 살펴보고 관련된 사업자간 정산문제에 대한 해결방안을 제시하고 있다.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.10-12
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• 2002

WWW의 급격한 팽창은 인터넷을 사용하는 사용자들의 급격한 증가를 가져 왔으며, 인터넷 사용자의 증가로 IP Address의 부족문제가 발생하게 되었다. 이에 대한 해결책으로 서브넷, 사설 IP Address, Proxy Server, NAT등의 다양한 해결책들이 제시되어 부분적으로 적용하고 있으나, 사설 IP Address를 사용하는 경우 인터넷 서비스를 이용하는 것은가능하나 인터넷 서버로서의 역할을 수행할 수 없다. 본 논문에서 설계 구현한 Gate-P 시스템은 사설 IP Address를 가지는 시스템을 인터넷 서버로서 활용할 수 있도록 한다.

• Journal of KIISE:Information Networking
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• v.35 no.5
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• pp.415-424
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• 2008

In this paper, we compare the QoS performance enhancement schemes according to the network architecture of mobile WiMAX system in order to improve the end-to-end QoS performance and propose QoS parameter mapping method and IP packet scheduling algorithm. To evaluate the end-to-end QoS performance, we implemented an end-to-end simulator of mobile WiMAX system using OPNET. Simulation results show that the proposed QoS parameter mapping scheme reduces the average delay of VoIP packet and improves uplink resource efficiency. And also, when the proposed IP packet scheduling algorithm is applied to the system, the end-to-end packet transmission delay of VoIP service can be reduced by 44-67 percent compared to FIFO and WRR scheduler.

• KIISE Transactions on Computing Practices
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• v.20 no.12
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• pp.688-693
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• 2014

With the advent of ubiquitous environments, the smart phone has come into wide use and the demand for various content increases. Thus, in order to efficiently utilize limited resources cognitive radio technology is regarded as a possible solution. Besides spectrum sensing or access schemes, the provision of VoIP traffic service for secondary users with limited spectrum resources is a very important issue. In this paper, a resource allocation and control system for VoIP QoS provision in cognitive radio networks is proposed. Firstly, as the system model, the time structure of the network is addressed and, according to the structure, a bandwidth broker is proposed. In addition, based on available bandwidth estimated by the bandwidth broker, a connection admission control for secondary users is developed. It is demonstrated that the provision of VoIP QoS is greatly affected by channel utilization, the number of channels, and the length of timeslot.

• IEMEK Journal of Embedded Systems and Applications
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• v.5 no.1
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• pp.7-17
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• 2010

ITS(Intelligent Transport System) as things are used for Broadcast service using TDMB/TPEG/NAVI rather than personal seamless service. It is attaching weight to Traffic information gathering, Charging, Settlement service. This research is applied to improve DCCP(Datagram Congestion Control Protocol) which has function as protecting data and preserving message boundary. The improving method is like that we solve data trust in UDP because Connection and Transmission overhead in UDP is less than in TCP. We fix the data loss which is generated from unordered delivery section of IP base wireless service by using DCCP protocol. We guarantee of connection with OBE(On-Board Equipment) and reliance about transmission of data by complement to mapping table and multi-hoping. Finally, We evaluate the performance about transmission of IP based data. We constructed a test-bed near research center for this test.

• Proceedings of the Korean Society of Computer Information Conference
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• 2012.07a
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• pp.291-294
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• 2012

본 논문에서는 VoIP를 이용한 어플리케이션을 제안한다. 네트워크 프로토콜인 VoIP와 SIP를 이용하여 영상통화 어플리케이션을 제작하고자 한다. SIP 프록시서버를 이용하여 송/수신자간에 안드로이드 어플리케이션을 사용하여 음성은 물론 영상까지 통화 가능한 어플리케이션이다. 또한, 본 시스템에서는 더 나아가서 송/수신자간의 통화전에 사전에 계약한 광고주의 광고들을 보여줌으로써 광고수익을 창출 할 수 있는 시스템을 개발하고자 한다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2004.05b
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• pp.420-423
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• 2004

Advance of computer technique becomes efficient of business in recent years. It has become high-speed data transmission and large data transmission. Network and computer system need to increasingly security because advance of computer technique. So this paper analyzes IP Traceback system that prevent cyber attack as hacking and security vulnerability of network. And this paper design IP Traceback system that based on active network.

• International journal of advanced smart convergence
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• v.10 no.2
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• pp.59-63
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• 2021

In order to broadcast 8K UHD, domestic broadcasting is upgrading production, operation, and transmission technologies in a new way that has various forms such as terrestrial broadcasting, IPTV, mobile, and streaming services. In order to overcome the existing SDI transmission technology that has reached its limit, new transmission technologies using ICT-based convergence such as IT, Network, IOT, and Big Data are developing. Internet Protocol(IP) transmission technology is an IT-based core protocol that transcends various broadcasting infrastructures. Therefore, it can be said that the technology for converting the broadcasting system based on IP is a natural one for preparing for 8K UHD in the future. In this paper, we propose an IP transmission system that adopts a network that can transmit 8K with hardware of 12G Serial Digital Interface (SDI), in order to prepare for the future broadcasting system according to the development of ALL IP interface technology.

• Journal of Advanced Navigation Technology
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• v.17 no.4
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• pp.421-428
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• 2013

Voice over IP refers to technology that enables routing of voice conversations over the Internet or a TCP/IP network. VoIP communication costs cheaper than traditional analog phone. Phone calls can be made to anywhere / anyone: Both to VoIP numbers as well as people with normal phone numbers. VoIP protocol equipment available today follows the SIP standard. Older VoIP equipment though would follow H 323, MGCP, Megaco/H.248. A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the TCP/IP network. A SIP server is also referred to a Asterisk IP-PBX. A VoIP telephone, also known as a SIP phone or a softphone, allows the user to make phone calls to any softphone, mobile or PC by using App store. A VoIP telephone can be a simple software-based softphone. However, the SIP Server and the program is vulnerable to VoIP attacks. In this paper, eavesdropping attacks tested by using the Asterisk SIP server. Eavesdropping attacks and TLS security methods apply to VoIP system. TLS can be applied to determine whether the eavesdropping available for VoIP Environments.