• Journal of Internet Computing and Services
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• v.9 no.6
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• pp.37-48
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• 2008

VoIP service is a transmission of voice data using SIP protocol on IP based network, The SIP protocol has many advantages such as providing IP based voice communication and multimedia service with cheap communication cost and so on. Therefore the SIP protocol spread out very quickly. But, SIP protocol exposes new forms of vulnerabilities on malicious attacks such as Message Flooding attack and protocol parsing attack. And it also suffers threats from many existing vulnerabilities like on IP based protocol. In this paper, we propose a new Virtual Proxy Server system in front of the existed Proxy Server for anomaly detection of SIP attack and stateful management of SIP session with enhanced security. Based on stateful virtual proxy server, out solution shows promising SIP Message Flooding attack verification and detection performance with minimized latency on SIP packet transmission.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.2
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• pp.1070-1088
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• 2017

Recently, the applications of Internet of Things (IoTs) are growing rapidly. Wireless Sensor Network (WSN) becomes an emerging technology to provide the low power wireless connectivity for IoTs. The IPv6 over low-power wireless personal area networks (6LoWPAN) has been proposed by IETF, which gives each WSN device an IPv6 address to connect with the Internet. The transmission congestion in IoTs could be a problem when a large numbers of sensors are deployed in the field. Therefore, it is important to consider whether the WSN devices have be completely integrated into the Internet with proper quality of service (QoS) requirements. The Software Defined Network (SDN) is a new architecture of network decoupling the data and control planes, and using the logical centralized control to manage the forwarding issues in large-scale networks. In this research, the SDN-based 6LoWPAN Border Router (6LBR) is proposed to integrate the transmission from WSNs to Internet. The proposed SDN-based 6LBR communicating between WSNs and the Internet will bring forward the requirements of end-to-end QoS with bandwidth guarantee. Based on our experimental results, we have observed that the selected 6LoWPAN traffic flows achieve lower packet loss rate in the Internet. Therefore, the 6LoWPAN traffic flows classified by SDN-based 6LBR can be reserved for the required bandwidth in the Internet to meet the QoS requirements.

• Journal of Broadcast Engineering
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• v.13 no.5
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• pp.696-706
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• 2008

In this paper we show the performance of the network-adaptive high-definition scalable video streaming using QWLAN board with IEEE 802.11e MAC monitoring and control. Realtime collected MAC parameters are used to determine which video data is extracted for the predicted available bandwidth. To achieve performance, extraction through R-D is proposed instead of the standard video packet extraction. It is shown through experiments that streaming video quality can be enhanced by fast adaptation to network conditions by using the proposed method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6B
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• pp.538-547
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• 2004

Optical Burst Switched (OBS) networks usually employ one-way reservation by sending a burst control packet (BCP) with a specific offset time, before transmitting each data burst frame (BDF). Therefore, The quality of service may be degraded because contentions may lead to loss of BDFs. Especially, this phenomenon becomes more serious when burst size is longer. This necessitates an effective method of prevention to avoid burst collision in nodes. OBS networks can employ several methods to avoid such burst losses. One is that burst size is cut short to reduce burst loss probability during scheduling time. In this paper, we evaluate the burst generation and transmission using Sub-Carrier Multiplexting (SCM) in OBS networks. We propose an appropriate burst assembly architecture and transmission scheme based on SCM in OBS networks. The performance of SCM in OBS networks is examined in terms of number of Sub-Carriers per wavelength, burst loss probability, throughput, and total bandwidth of an optical fiber.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.2C
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• pp.188-195
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• 2012

Many systems are developing for the realization of NCW(Network Centric Warfare). UAV(Unmanned Aerial Vehicle) Network is attracting attention in a lot of military applications. In general, UAVs have the potential to create an ad-hoc network and greatly reduce the hops from source to destination. However, UAV networks exhibit unique properties such as high mobility, high data rate, and real time service. The routing protocols are required to design the multi-hop routing protocols that can dynamically adapt to the requirements of UAV network. In this paper we analyse Geographic Routing Protocol is based on geographical distance between source and destination for efficient and reliable transmission. Geographic Routing Protocol is evaluated in video service scenarios with TDMA model in our simulation. The simulation results show that the performance of Geographic Routing Protocol is better than the MANET Routing Protocol in terms of packet received ratio, end to end delay, and routing traffic sent.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.10B
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• pp.1043-1049
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• 2009

Broadcast of sink node is used for network management, data collection by query and synchronization in wireless sensor networks. Simple flooding scheme induces the broadcast storm problem. The MPR based broadcast schemes reduce redundant retransmission of broadcast packets. MPR is a set of one hop neighbor nodes which have to relay broadcast message to cover all two hop neighbors. Though MPR can reduce redundant retransmission remarkably, it still suffers from energy waste problem caused by collision and duplicate packets reception. This paper proposes a new MPR based sink broadcast scheme using 3-channel. The proposed scheme reduces energy consumption by avoiding duplicate packet reception, while increases reliability by reducing collision probability remarkably. The results of analysis and simulation show that the proposed scheme is more efficient in energy consumption compared to the MPR based scheme. The result also shows that the proposed scheme reduces delivery latency by evading a contention with other relay nodes and improves reliability of broadcast message delivery by reducing collision probability.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.10B
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• pp.865-875
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• 2008

In this paper, we propose a reliable multicast scheme for the tactical wireless ad hoc network where the members share the situation and operation data. Our scheme consists of two phases to guarantee the reliability. During the dissemination phase, we utilizes an epidemic approach similar to gossip-based ad hoc routing. Then the group members interchange the message reception state information periodically so that they are able to recover missing packets from the other members. We evaluate the performance of our scheme through extensive simulations using the network simulator in comparison with other existing reliable multicast schemes in mobile ad hoc network. Our scheme shows higher packet delivery ratio regardless of the mobility and imposes much lower control overhead to the non-group members compared with the existing schemes.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.7B
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• pp.583-591
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• 2008

This paper proposes a resource allocation algorithm for minimizing the overall distortion of multiple users in orthogonal frequency division multiplex (OFDM). The proposed algorithm exploits the diversity of multiuser and the rate-distortion function using packet distortion model in a system with limited resources. We first induce a rate-distortion function considering error concealment and error propagation properties of H.264 video structures. Then we perform adaptive resource allocation utilizing multiuser diversity for minimizing the overall video quality degradation. We also consider the proportional rate which is pre-determined for each user. Simulation results show that compared to the previous time division multiple access method and the resource allocation method maximizing data rate, the proposed rate allocation algorithm substantially improves the received video quality.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.3
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• pp.101-108
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• 2009

Since the use of wireless communication instruments was standardized, users expect to be provided with seamless information whenever and wherever they use the instruments. Also, some technology is required to satisfy the users' needs which will cover their mobility. To support the mobility of host, the Internet Engineering Task Force (lETF) Mobile IP Working Group proposed a protocol called MIPv6 (Mobile IPv6). But in the case of the existing MIPv6, sometimes Mobile Node cannot receive data packet if Handover occurs although it is a temporal phenomenon. For solving these Handover problems, there are many methods like FMIPv6 (Fast Handover for Mobile IPv6) and HMIPv6 (Hierarchical Mobile IPv6) have been suggested. This paper suggested the use of Dual Buffer of Access Point and an effective way of registration as a way of reducing delayed time caused by Handover. Also, it analyzed and compare the existing MIPv6 with a proposed scheme concerning delayed time of Handover. Finally, the main objective of this paper is to proposed scheme that can reduce the delayed time of Handover compare to the existing MIPv6.