• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.6
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• pp.63-69
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• 2010

Today, Internet Telephony Services based on VoIP are gaining tremendous popularity for general user. Therefore a various demands of the user keep up increase, the most important requirements of these is voice security about telephony system. It is needed to ensure secret of voice call in a special situation. Due to the fact that many users can connect to the internet at the same time, VoIP can always be in a defenseless state by hackers. Therefore, in this paper, we have developed VPN IP-PBX for the voice security and measured conversation quality by adopting VPN IPsec based on SIP and using tunnel method in transmitting voice data to prevent eavesdrop of voice data. This VPN IP-PBX that is connected Soft-phone provide various optional services.

• Journal of Internet Computing and Services
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• v.6 no.1
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• pp.13-25
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• 2005

Distributed Denial-of-Service(DDoS) attack prevent users from accessing services on the target network by spoofing its origin source address with a large volume of traffic. The objective of IP Traceback is to determine the real attack sources, as well as the full path taken by the attack packets. Existing IP Traceback methods can be categorized as proactive or reactive tracing. Existing PPM based tracing scheme(such as router node appending, sampling and edge sampling) insert traceback information in IP packet header for IP Traceback. But, these schemes did not provide enhanced performance in DDoS attack. In this paper, we propose a 'TTL based advanced Packet Marking' mechanism for IP Traceback. Proposed mechanism can detect and control DDoS traffic on router and can generate marked packet for reconstructing origin DDoS attack source, by which we can diminish network overload and enhance traceback performance.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.12
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• pp.2555-2562
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• 2011

Binary CDMA is a new standard technology for wireless communication developed by our country that makes high speed communications and good quality of services. In this paper we researched the development methods and implementation of a wireless VoIP System based on the Binary-CDMA technology that makes it freely installed in any place without phone cables and laying works. We designed the structure of the Binary-CDMA Wireless VoIP System and defined messages and data structures for call control. And then we designed and implemented call control message flow between VoIP terminals for user's communication.

• The KIPS Transactions:PartC
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• v.15C no.6
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• pp.567-572
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• 2008

It is necessary to study about the scheme to accept IP traffic effectively in OTN, according to expect that IP traffic data will be increase constantly. In this paper, we studied the encapsulation method of IP traffic through GFP in OTN transmission network. Therefore we knew the IP/GFP/OTN method is more efficient than existing methods from structure, overhead rates and possibility of grow, and showed the functional block of high level about this method. Also we showed the implementation scheme of this method by processing the functional simulation to make use of VHDL programming.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.1
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• pp.156-166
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• 2010

Development of IP network technology creates new service which applying to fusion technologies such as IPTV, VoIP, and so on. Especially, Next generation IP is called IPv6 which will solve the lack of IP. IPv6 is very important on IPTV which needs best quality of service about security, QoS, and bandwidth. In this paper, we constructed experimental network, measured PSNR which computes the peak signal-to-reconstructed video measurement in experimental network environment, analyzed PSNR value. And as a result of measurement, we, Propose optimum network environment for IPTV service provider and transition mechanism designer.

• The Journal of the Korea institute of electronic communication sciences
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• v.6 no.1
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• pp.49-54
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• 2011

In this paper, End-to-End VoIP(Voice over Internet Protocol) performance over large scale MANET(Mobile Ad-hoc Networks) is evaluated. VoIP implementation model, which can be used for large scale MANET, is suggested through this evaluation. VoIP simulator based on NS-2 network simulator is used for performance measurement. Call connection ratio as a connection performance, MOS and network delay as a transmission performance, packet loss rate as other performance is measured with this simulation. Finally, maximum $10{\sim}15km^2$ as a single MANET, 500 nodes and 100 call connections is suggested for VoIP implementation and operation conditions over large scale MANETs.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.5
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• pp.95-100
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• 2011

The next generation IP-PBX system are demanding multimedia facility to carry out UC(Unified Communication) and voice security also. Therefore, in this paper, we have developed the integrated solution of IP-PBX for the voice security by adopting VPN IPsec based on SIP using tunnel method in transmitting voice data to prevent eavesdrop of voice data and have shared between communication system based on PC and PSTN terminals. In particular, We have developed a video conference, private switching, distributed processing and measured telephone conversation quality. This IP-PBX that is connected Soft-phone provide various optional services.

• Journal of Korea Society of Digital Industry and Information Management
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• v.7 no.1
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• pp.31-39
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• 2011

VoIP service uses packet network of ip-based because that has eavesdropping, interception, illegal user as vulnerable elements. In addition, PSTN of existing telephone network is subordinate line but VoIP service using the ip packet provide mobility. so The user authentication and VoIP user's account service using VoIP has emerged as a problem. To solve the vulnerability of SIP, when you use VoIP services with SIP, this paper has made it possible to authenticate user's terminal by using proxy server and proxy server by using authentication server. In conclusion, sender and receiver are mutually authenticated. In the mutual authentication process, the new session key is distributed after exchanging for the key between sender and receiver. It is proposed to minimize of service delay while the additional authentication. The new session key is able to authenticate about abnormal messages on the phone. This paper has made it possible to solve the vulnerability of existing SIP authentication by using mutual authentication between user and proxy server and suggest efficient VoIP service which simplify authentication procedures through key distribution after authentication.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.10
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• pp.2164-2170
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• 2011

Rapid surge in date quantity lead to increase in storage demand from corporate. The existing SAN with fiber channel is being changed to IP-based SAN environment due to installment and maintenance cost. But the IP-based network still have some similar security problems as existing TCP/IP network. Also, for the security reasons of storage traffic, data are encrypted, but with the existing system, data larger than 10G can't be handled. To address security and speed issue, this paper proposes to a structure applied to IP SAN environment with Parallel Structure Word-based FSR (PS-WFSR) as hardware.

• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.2
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• pp.93-100
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• 2009

As the Internet Protocol be widely/rapidly used in packet communication, common carriers are providing the multimedia service(Both direction real-time voice, video conference, remote educational etc.)on the Internet. Also the 070 VoIP (Voice over IP) service is provided by the carriers on the packer network. In order to offer VoIP service in Korea, common carrier has to acquire the optimum level for QoS(quality of service). In this paper, we study on CODEC quality to get a higher QoS for VoIP.