• Journal of KIISE:Information Networking
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• v.27 no.2
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• pp.197-205
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• 2000

This paper proposes and evaluates a scheme which enhances real-time performance on the Ethernet-based residential network where cable length is very short and most of traffic passes through a specific device. The scheme aims at enhancing deadline meet ratio of the real-time traffic, for example, video stream by means of applying the functionalities such as queue discrimination and early packet discard as well as maintaining network load properly. The design includes the description on hardware interfaces along with the software modules such as scheduler, device driver and connection manager. The simulation results via SMPL suggest that the proposed network can be used as a cost-effective residential network. We also measure the permissible bandwidth for non-real-time traffic without degrading the performance of real-time traffic.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.2
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• pp.40-49
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• 2015

PicoCast is a recently proposed short-range wireless communications technology that supports both low rate sensor/control data and high speed data such as voice and video traffic in the personal space, defined by the user-centric space of radius tens of meters, with one unified protocol. When the users are in the same area, personal spaces defined by each user may overlap. Since these PicoCast piconets may simultaneously operate, mutual interference is unavoidable. It is necessary to investigate the effect of mutual interference on data transmission and to conduct research on minimizing the interference among PicoCast devices. In this paper we analyze the effect of mutual interference between PicoCast piconets based on data throughput using OPNET network simulator. We have implemented the PicoCast protocol and measured the average throughput and packet loss rate.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.12 no.3
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• pp.96-102
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• 2013

In recent years, a variety of ITS services are available such as driving information, road conditions, V2X messages as well as navigation and traffic jams notification. The development of ITS services is accelerating by V2X communication technologies for high-speed vehicles. In this paper, WAVE communication devices based on the IEEE802.11p standard is introduced as a solution of V2X communication technologies. The H/W and S/W structures of the WAVE communication device and the characteristics of RF/antenna are described. The performance is evaluated in the test road by measuring throughput, PER and latency. The implemented WAVE communication device has 6~7 Mbps throughput with 10% PER at 1km coverage. The packet latency is less than 3ms for the whole test road. It is shown that the implemented WAVE technology is satisfactory to provide ITS services and Internet video-streaming services.

• The KIPS Transactions:PartC
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• v.11C no.7 s.96
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• pp.871-878
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• 2004

Recenty, there has been great interest in MANET from various areas. In this paper we focus on performance analysis of on demand routing protocols surf as DSR, AODV, and TORA in MANET. We have conducted several simulations concerned with application service data such as sensor, text. voice, and video data. And then, we have evaluated the performance of three protocols in a pre-designed ad hoc network, which is consisted of 20 nodes. As a result, we have obtained quantitative data for packet delivery fraction, average end to-end delay, routing load, channel utilization from upper layer and supportable routing protocol for application service data. The results can be used for designing specific-purposed ad-hoc networks.

• 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.32 no.4B
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• pp.227-233
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• 2007

According to the recent development of network technology, the demands of users are diversified and the needs of multimedia traffic are increasing. In general, UDP(User Datagram Protocol) traffic is used to transport multimedia data, which satisfied the real-time and isochronous characteristics. UDP traffic competes with TCP traffic and incur the network congestion. However, TCP traffic performs network congestion control but does not consider the receiver's status. Thus, it is not appropriate in case of streaming services. In this paper, we solve a fairness problems and proposed a network algorithm based on RTP/RTCP(Real-time Transport Protocol/Realtime Transport Control Protocol) in view of receiver status. The POBA(Priority Ordered Buffer Algorithm), which applies priorities in the receiver's buffer and networks, shows that it provides the appropriate environment for streaming services in view of packet loss ratio and buffer utilization of receiver's buffer compared with the previous method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.3B
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• pp.437-445
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• 2000

More advanced and demanding applications, like videoconferencing, video on demand. distributed computing etc., have been devised thanks to the availability of enhanced network feature. Network technologies need to be enhanced to support these applications and to cope with the increasing number of users. Increasing the availability of network resources is just not enough to achieve this goal; scalable network architecture. increased packet forwarding capabilities, a wider range of services are all additional requirements. MOLS (multiprotocol label Switching) is one of the new networking techniques under standardization in the scientific community. In this paper, we analyze the performance of label switching in ATM networks. Simulation tests were performed. In this simulation, we set a simple network configuration and used internet traffic traces from NLANR as input traffic sources. The simulation results and analysis will be helpful in utilizing the functions of ATM switching and IP routing.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.1C
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• pp.37-47
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• 2009

When H.264 encoded video streams are transmitted over wireless network, packet loss is unavoidable. Responding on this environment, we propose methods to recover missed motion vector in the decoder: At first, A candidate vector set for missing macroblock is estimated from high correlation coefficient of neighboring motion vectors and missing block vectors the algorithm clusters candidate vectors through distances amongst motion vectors of neighboring blocks. Then the optimal candidate vector is determined by the median value of the clustered motion vector set. In next stage, from the candidate vector set, the final candidate vector of missing block is determined it has minimum distortion value considering directions of neighboring pixels' boundary. Test results showed that the proposed algorithm decreases the candidate motion vectors $23{\sim}61%$ and reduces $3{\sim}4sec$ on average processing(decoding) time comparing the existing H.264 codec. The PSNR, in terms of visual quality is similar to existing methods.

• The KIPS Transactions:PartC
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• v.19C no.1
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• pp.71-82
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• 2012

In Next Generation Wireless Networks (NGWN), lots of information, such as voice and video data, will be used in All-IP networks. It is important note that seamless service for users and handoff between heterogeneous networks must be taken into account in such environments. Therefore, in this paper, we propose a SePH(Seamless Proxy based Handoff) using PMIPv6-based proxy model which is enable to improve the performance of handoff in NGWN. The SePH can efficiently support seamless and IP-based mobility by reducing the search process. The performance results show that our proposed scheme outperforms in terms of QoS (Quality of Service) such as throughput, handoff latency, packet loss, and signaling overhead comparing to the existing schemes.

• Journal of Korea Multimedia Society
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• v.7 no.5
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• pp.668-679
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• 2004

When users of mobile environments move fast or slow into a number of base stations(BS) and request the services of continuous media data such as video or audio, this study examines what the caching has the usefulness in mobile environments. Namely, to reduce packet disconnections and network overheads in mobile environments and minimize transmission delay time, we propose domain-based hierarchical caching structure and study whether application of caching has the usefulness. So we have a model based on user environments and hierarchical network structure to process continuous media services, and analyze the usefulness of caching which depends on the mobile patterns of user and the locations of caching nodes. And then, we research whether caching offers the usefulness in mobile environments. As the result, we are able to see that an adaptable application of caching is needed because the hit ratio and the number of replacement vary in large according to mobile patterns of user and locations of caching.