• Proceedings of the Korean Information Science Society Conference
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• 2007.10b
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• pp.429-434
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• 2007

프로세스의 우선순위는 임베디드 시스템에서 수행되는 여러 가지 작업들의 중요도 또는 작업마감 시간의 임박성에 따라 결정되기 때문에 프로세스의 입출력 작업 시에도 반영되어야 한다. 하지만 많은 임베디드 운영체제들은 기존의 범용 운영체제를 기반으로 설계되었기 때문에 입출력 작업 시 프로세스의 운선순위를 반영하지 못하고 있다. 본 논문에서는 이러한 문제를 해결하기 위해서 새로운 통신 프로토콜 스택 구조를 제안하고 이를 임베디드 리눅스에 구현한다. 또한 본 논문은 이더넷이 산업용 기기 등의 연결에 활용될 수 있음에 주목하고 독립 이더넷 네트워크에 적합한 전송 프로토콜을 제안한다. 측정 결과 제안된 프로토콜 스택 RTDiP($\underline{R}eal-\underline{T}ime\;\underline{Di}rect\;\underline{P}rotocol$)은 UDP/IP와 비교하여 단방향 통신 지연시간을 최대 59% 감소시켰으며 통신처리율을 최대 155% 향상시킬 수 있음을 보인다. 또한 낮은 우선순위를 갖는 배경 통신 프로세스에 의해서 UDP/IP는 532%가 단방향 통신 지연시간이 증가하나, RTDiP은 2% 미만의 증가만을 보임으로써 프로세스의 우선순위에 따라 패킷 처리가 이루어지고 이를 통해서 실시간 통신을 지원해줄 수 있음을 보인다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2012.05a
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• pp.667-669
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• 2012

Traditional DDoS defence methods are performed at server side which was attacked. If servers detect DDoS attack, they use some methods for defending the attack such as increasing the bandwidth, bypassing the traffic, blocking the IP addresses or blocking the ports by the firewall. But as lots of people use smart-phones, it is possible a smart-phone to be a zombie and DDoS attack could be much more a huge and powerful forms than now. Victims are not only a server but also a host which becomes a zombie. While it performs DDoS attack, zombie smart-phone users have to pay the extra charge. After finish the attack, DDoS try to destroy hard drives of zombie hosts. Therefore the situation is changed rather than to defend DDoS server side only, we should protect a client side who needs to prevent DDoS attacks. In this paper, we study a defence method that we terminates a process which perform the attack, send the information to different hosts when a zombie PC or smart-phone perform DDoS attacks.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.8
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• pp.3728-3752
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• 2016

Wireless mesh networks (WMNs) have been increasingly applied in private and public networks during the last decade. In a different context, voice over IP (VoIP) has emerged as a new technology for making voice calls around the world over IP networks and is replacing traditional telecommunication systems. The popularity of the two technologies motivated the deployment of VoIP over WMNs. However, VoIP over WMNs suffers from inefficient bandwidth utilization because of two reasons: i) attaching 40-byte RTP/UDP/IP header to a small VoIP payload (e.g., 10 bytes) and ii) 841 μs delay overhead of each packet in WMNs. Among several solutions, VoIP packet multiplexing is the most prominent one. This technique combines several VoIP packets in one header. In this study, we will survey all the VoIP multiplexing methods over WMNs. This study provides a clear understanding of the VoIP bandwidth utilization problem over WMNs, discusses the general approaches in which packet multiplexing methods could be performed, provides a detailed study of present multiplexing techniques, shows the aspects that hinder the VoIP multiplexing methods, discusses the factors affected by VoIP multiplexing schemes, shows the merits and demerits of different multiplexing approaches, provides guidelines for designing a new improved multiplexing technique, and provides directions for future research. This study contributes by providing guidance for designing a suitable and robust method to multiplex VoIP packets over WMNs.

• Journal of Korea Multimedia Society
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• v.10 no.3
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• pp.335-347
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• 2007

VoIP (Voice over IP) is a technology to transport video and voice traffic over IP networks such as Internet. Today, the VoIP technology is viewed as the right choice for providing voice, video, and data communication among various terminals over the next generation network. This paper discusses a multipoint VoIP implementation with end-point mixing model which can support multipoint conference without a conference bridge. The multipoint VoIP is implemented with SIP (Session Initiation Protocol), and supports STUN (Simple Traversal of UDP Through NATs) since it works in an asymmetric NAT (Network Address Translator) environment. The characteristics of this paper are as follows. It is possible that all terminals in the hierarchical conference don't receive the duplicated media information because we use the end-point mixing model with the new media processing module. And, the paper solves the problem that the hierarchical conference session should be separated into several sessions when a mixing terminal terminates the hierarchical conference session.

• Journal of Broadcast Engineering
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• v.9 no.2
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• pp.119-130
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• 2004

We have developed a prototype super-high-definition (SHD) digital cinema distribution system that can store, transmit, and display eight-million-pixel motion pictures that have the image quality of a 35-mm film movie. The system contains a movie server, a real-time decoder, and an SHB projector. Using a Gigabit Ethernet link and TCP/IP, the server transmits JPEG2000 compressed motion picture data streams to the decoder at transmission speeds as high as 300 Mbps. The received data streams are decompressed by the decoder, and then projected onto a screen via the projector. By using an enlarged TCP window, multiple TCP streams, and a shaping function to control the data transmission quantity, we achieved real-time streaming of SHD movie data at about 300 Mbps between Chicago and Los Angeles, a distance of more than 3000 km. We also improved the decoder performance to show movies with Image qualities of 450 Mbps or higher. Since UDP is more suitable than TCP for fast long-distance streaming, we have developed an SHD digital cinema UDP relay system, in which UDP is used for transmission over a fast long-distance network. By using four pairs of server-side-proxy and decoder-side-proxy, 450-Mbps movie data streams could be transmitted.

• Proceedings of the Korea Information Processing Society Conference
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• 2003.05b
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• pp.1353-1356
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• 2003

본 논문에서는 EP 보안(IPsec : IP Security)이 적용된 보안 패킷들을 네트워크 상에서 실시간으로 수집하여 분석해 주는 IP 보안 패킷 분석기를 설계 및 구현하였다. 본 패킷 분석기는 TCP UDP IP, ICMP 등의 일반 네트워크 패킷과 키 교환을 위한 IKE 패킷, 보안 통신을 위한 AM, ESP 패킷 등을 실시간으로 수집하고 분석하는 기능을 갖는다. 본 패킷 분석기는 현재의 IPv4 패킷 뿐 아니라 차세대 인터넷인 IPv6 패킷에 대하여도 실시간 수집 및 분석 기능을 제공한다. 또한 본 분석기는 IPsec 엔진에 대한 보안성을 평가하기 위한 자동화된 평가기능도 제공해 준다. 개발한 패킷 분석기를 이용하여 ETRI에서 개발한 통합 IPsec 엔진에 대한 보안성을 평가한 결과도 함께 보인다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.11A
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• pp.1051-1058
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• 2010

Mobile IP (MIP) and SIP are considered as important technologies to provide the macro mobility in the next generation mobile convergence networks which have heterogeneous access networks. Typically, MIP and SIP are more suitable for the non-real-time TCP connections and the real-time RTP/UDP sessions respectively, hence a handset which uses both of these sessions should simultaneously apply MIP and SIP to perform the efficient mobility management. Existing multi-layered mobility management schemes focus on the signalling order of each protocol. However, simple combining of two protocols cannot provide the performance enhancement of the mobility management. In this paper, a novel multi-layered mobility management algorithm using the combination of SIP and fast MIPv6 (FMIPv6) is proposed. FMIPv6 and SIP mobility is simultaneously performed to reduce the service interrupt time and to guarantee QoS requirement. The delay model is defined to analysis the performance of the algorithm and the simulation results show the performance of the proposed algorithm.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.3
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• pp.111-119
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• 2010

With the rapid progress of wireless technologies today, mobile terminals with multiple access interfaces are emerging. In recent years, WLAN (Wireless Local Area Networks) has become the premier choice for many homes and enterprises. WiMAX (Worldwide Interoperability for Microwave Access) has also emerged as the wireless standard that aims to deliver data over long distances. Therefore, it is important to explore efficient integration methods for delivering multimedia data between heterogeneous wireless networks. In this paper, we developed the simulation models and environments for the mobile multihomed node that has both WiMAX and WLAN interfaces and can move around in both networks by using mobile IP. In order to verify the developed models, we designed and constructed several simulation scenarios, e.g. movement in WiMAX/WLAN, group mobility, MANET, and nested MIP under the various traffic environments such as oneway or bothway UDP packets, FTP traffic, and voice with SIP protocol. The simulation results show that the developed models are useful for mobility studies in various integrated wireless networks.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.9
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• pp.373-383
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• 2003

In this paper we discuss the effect of the delay limit and the packet size related to the quality of service on a VoIP system using the Internet. We also provide a guideline to determining the optimal packet size of the voice data for a given delay limit. Empirical studies are done with two personal computers connected through the packet switched public IP network. The sender encodes the voice signal from the microphone to get PCM and ADPCM data and sends the data to the receiver using UDP packets. The receiver plays the reconstructed voice from the stream with lost and delayed packets. The quality of the reconstructed voice is evaluated offline by the MNB (Measuring Normal Block) method using the data acquired from the both sides. The result shows that under the delay limit of 100ms for 40Kbps, 32Kbps and l6Kbps of ADPCM data, the minimum packet size should be 300bytes, 400bytes and 600bytes respectively and the maximum packet size should be l200bytes commonly for the best quality of voice.

• The KIPS Transactions:PartC
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• v.9C no.2
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• pp.235-246
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• 2002

While the Internet is suffering from the massive data such as video stream, IP multicast can ease the load of the Internet by enabling one copy of digital information to be received by multiple computers simultaneously. But If multicast is based on UDP, packets are delivered using a best-effort Policy without any reliability, congestion control or flow control. Multicast group members can join or leave a multicast group at will, and multicast uses broadcast mechanism, it's very hard to keep security from unauthorized members. In this paper, we introduce a new reliable multicast protocol TRDMF proper for one-to-many multicast model with reliability, flow control, congestion control and key management.