• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.05a
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• pp.704-711
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• 2003

Recently, the demands for real-time service and multimedia data are rapidly increasing. There are significant redundancies between header fields both within the same packet header and in consecutive packets belonging to the same packet stream. But there are many overheads in using the current UDP/IP protocol. Header compression is considered to enhance the transmission efficiency for small size of payload. By sending the static field information only once initially and by utilizing dependencies and predictability for other fields, the header size can be significantly reduced for most packets. This work describes an implementation for header compression of the headers of U/UDP protocols to reduce overhead on Ethernet network. Typical UDP/IP Header packets can be compressed down to 7 bytes and the header compression system is designed and implemented on the Linux environment. Using the designed Header compression system between a server and a client have the advantage of effective data throughput in network.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.5
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• pp.979-985
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• 2016

In this paper, a distributed instant message system with multiple servers architecture which can distribute system load effectively using an extended media control channel has been presented. A media control channel provides establishing a reliable control channel and also keeping a reliable control channel between SIP server and client in the field of real-time media transport area. In this study, a new instant message system architecture which can distribute massive instant message including multimedia data to multiple servers has been presented. The presented instant message system architecture can distribute system load by extending media control channel. For this purpose, media control channel messages, which distribute system load to multiple servers dynamically according to increasing number of users, have been designed in our presented system. And, in our research, an exchanging procedures of media control channel messages between servers have also been presented. The performance of the proposed system has been analysed by simulation.

• Journal of KIISE:Computer Systems and Theory
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• v.35 no.9_10
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• pp.466-475
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• 2008

Owing to the improved wireless communication technologies, it is possible to provide streaming service of multimedia with PDAs and mobile phones in addition to desktop PCs. Since mobile client devices have low computing power and low network bandwidth due to wireless network, the transcoding technology to adapt media for mobile client devices considering their characteristics is necessary. Transcoding servers transcode the source media to the target media within corresponding grades and provide QoS in real-time. In particular, an effective load balancing policy for transcoding servers is inevitable to support QoS for large scale mobile users. In this paper, the transcoding load estimation algorithm is proposed for load balance on the distributed transcoding environments. The proposed algorithm estimates transcoding time from transcoding server information, movie information and target transcoding bit-rate. The estimated transcoding time is proved based on experiments.

• The KIPS Transactions:PartD
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• v.12D no.2 s.98
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• pp.303-308
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• 2005

Generally, the overloaded server problem and the rapidly increased network traffic problem are happened in a center concentrated multimedia digital content service. Recently, a study about the CDN which is a digital content transmission technology to solve these problems are performed actively. In this study, we proposed the SDCDS which improved a process latency time and a security performance on a digital content delivery and management. The goal of the SDCDS is the digital content security and the improvement of the processing time. For that, we have to design the security and the caching method considering the architecture characteristics of the CDN. In the SDCDS, the public key encryption method is designed by considering the architecture characteristics of CDN. And we improved the processing latency time by improved the caching method which uses the grouped caching method on the encrypted DC and the general DC. And in the experiment, we veryfy the performance of the proposed system.

• The KIPS Transactions:PartC
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• v.13C no.2 s.105
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• pp.203-210
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• 2006

Fair packet scheduling algorithms supporting quality-of-services of real-time multimedia applications can be classified into the following two design schemes in terms of the reference time used in calculating the timestamp of arriving packet: Finish-time Design (FD) and Start-time Design (SD) schemes. Since the former can adjust the latency of a flow with raising the flow's reserved rate, it has been applied to a router for the guaranteed service of the IETF (Internet Engineering Task Force) IntServ model. However, the FD scheme may incur severe bandwidth loss for traffic flows requiring low-rate but strong delay bound such as internet phone. In order to verify the usefulness of the SD scheme based router for the IETF guaranteed service, this paper analyzes and compares two design schemes in terms of bandwidth and payload utilizations. It is analytically proved that the SD scheme is better bandwidth utilization than the FD one, and the simulation result shows that the SD scheme gives better payload utilization by up to 20%.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.12
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• pp.51-60
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• 2004

DAB(Digital Audio Broadcasting) is a new media service that can provide CD quality audio, various data service, interactive and high quality mobile communications through popular media such as terrestrial broadcasting, satellite, cable TV, and internet. In this paper, a new EPG(Electronic Program Guide) application model is proposed. The model is based on DAB and combines a DAB receiver and PCs so that it can take advantages of using various multimedia services and plenty of internet contents. The developed EPSD(EPG Provider System on DAB) has Web-based Server/Client structure and povides EPG functionalities to client PCs over internet. Therefore, the DAB receiver can be smaller and cheaper, and can develop abundant data services on internet. It can also provide high quality video services and be expected to become an important component in future home network systems.

• The KIPS Transactions:PartC
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• v.11C no.7 s.96
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• pp.951-958
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• 2004

Nowadays NGN is developed for supporting the e-Commerce, Internet trading, e-Government, e-mail, virtual-life and multimedia. Internet gives us the benefit of remote access to the information but causes the attacks that can break server and modify information. Since 2000 Nimda, Code Red Virus and DSoS attacks are spreaded in Internet. This attack programs make tremendous traffic packets on the Internet. In this paper, we designed and developed the Bandwidth Controller in the gateway systems against the bandwidth saturation attacks. This Bandwidth con-troller is implemented in hardware chipset(FPGA) Virtex II Pro which is produced by Xilinx and acts as a policing function. We reference the TBF(Token Bucket Filter) in Linux Kernel 2.4 and implemented this function in HDL(Hardware Description Language) Verilog. This HDL code is synthesized in hardware chipset and performs the gigabit traffic in real time. This policing function can throttle the traffic at the rate of band width controlling policy in bps speed.

• Journal of Internet Computing and Services
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• v.5 no.1
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• pp.15-23
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• 2004

This paper presents distributive web-caching technique on the Head-End-Network (HNET) to solve excessive load of multimedia sever and inefficient use of network resources, The proposed web-caching is not concentration of load for a specific Head-End-Node(HEN), and it has advantage that there is no copy of a specific item on distributive HENs. This technique distributively stores on HENs partial streams of requested videos from clients connected to HENs and the order of store streams follows the order of request of identical video items from HENs. Thus, storing streams on each HEN that requests sevice are different. When a client requests the cached video on some HENs, the HEN that connects him receives streams in the order of store from HENs which stored them and it transmits them to him. These procedures are performed under the control of SA. This technique uses cache replacement algorithm that the combination of LFU, LRU and the last remove for the first stream of videos In order to reduce end-to-end delay.

• Journal of Internet Computing and Services
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• v.9 no.2
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• pp.83-88
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• 2008

Session initiation protocol (SIP) is an application-layer prolocol to initiate and control multimedia client session. When client ask to use a SIP service, they need to be authenticated in order to get service from the server. Authentication in a SIP application is the process in which a client agent present credentials to another SIP element to establish a session or be granted access to the network service. In 2005, Yang et al. proposed a key exchange and authentication scheme for use in SIP applications, which is based on the Diffie-Hellman protocol. But, Yang et al.'s scheme is not suitable for the hardware-limited client and severs, since it requires the protocol participant to perform significant amount of computations (i.e., four modular exponentiations). Based on this observation. Huang and Wei have recently proposed a new efficient key exchange and authentication scheme thor improves on Yang et al.'s scheme. As for security, Huang and Wei claimed, among others, that their scheme is resistant to offline dictionary attacks. However, the claim turned out to be untrue. In this paper, we show thor Huang and Wei's key exchange and authentication scheme is vulnerable to on offline dictionary attack and forward secrecy.

• 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.