• The KIPS Transactions:PartC
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• v.12C no.3 s.99
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• pp.429-436
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• 2005

A mobile ad hoc network (MANET) is a collection of mobile nodes without any fixed infrastructure or my form of centralized administration such as access points and base stations. The ad hoc on-demand distance vector routing (AODV) protocol is an on-demand routing protocol for MANETs, which is one of the Internet-Drafts submitted to the Internet engineering task force (IETF) MANET working group. This paper proposes a new multipath routing protocol called maximally disjoint multipath AODV (MDAODV), which exploits maximally node- and link-disjoint paths and outperforms the conventional multipath protocol based on AODV as well as the basic AODV protocol. The key idea is to extend only route request (RREQ) message by adding source routing information and to make the destination node select two paths from multiple RREQs received for a predetermined time period. Compared to the conventional multipath routing protocol, the proposed MDAODV provides more reliable and robust routing paths and higher performance. It also makes the destination node determine the maximally node- and link-disjoint paths, reducing the overhead incurred at intermediate nodes. Our extensive simulation study shows that the proposed MDAODV outperforms the conventional multipath routing protocol based on AODV in terms of packet delivery ratio and average end-to-end delay, and reduces routing overhead.

• The KIPS Transactions:PartC
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• v.13C no.4 s.107
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• pp.471-480
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• 2006

Recently, multimedia communication services, such as video conferencing and voice over IP, have been rapidly spread. H.323 is an international standard that specifies the components, protocols and procedures that provide multimedia communication services of real-time audio, video, and data communications over packet networks, including IP based networks. H.323 is applied to many commercial services because it supports various network environments and has a good performance. But communication services based on H.323 may have some problem because of current network trouble or mis-implementation of H.323. The understanding of this problem is a critical issue because it improves the quality of service and is easy to service maintenance. In this paper, we implement the analysis system for H.323 protocol wihch includes H.245, H.225.0, RTP, RTCP, and so on. Tills system is able to capture, parse, and present the H.323 protocol in real-time. Through the operation test and performance evaluation, we prove that our system is a useful to analyze and understand the problems for communication services based on H.323.

• Journal of Korea Multimedia Society
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• v.11 no.7
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• pp.1007-1015
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• 2008

In order to provide real-time data from sensors and instruments at manufacturing processes on web, we proposed a communication service model based on XML(eXtensible Markup Language). HTML(Hyper Text Markup Language) is inadequate for describing real-time data from manufacturing plants while it is suitable for display of non-real-time multimedia data on web. For applying XML-based web service of process data in Intranet environment, real-time performance of communication services was evaluated to provide the system design criteria. XML schema for the data presentation was proposed and its communication performance was evaluated by simulation in terms of transmission delay due to increased message length and processing delay for transformation of raw data into defined format. For transformation of raw data into XML format, we proposed two structures: one is the scheme where transformation is done at an SCC(Supervisory Control Computer) after receiving real-time data from instruments. the other is the scheme where transformation is carried out at instruments before the data are transmitted to the SCC. Performances of two structures were evaluated on a testbed under various conditions such as six packet sizes and offered loads of 20%, 50% and 80%, respectively. Test results show that proposed schemes are applicable to the systems in Ethernet 100BaseT network if total message traffic is less than 7 Mbps.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.7
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• pp.63-73
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• 2015

In the end to end data transfer protocols, it is very important to correctly estimate available bandwidth. In UDT (UDP based Data Transfer), receiver estimates the MTR (Maximum Transfer Rate) of the current link using pair packets transmitted periodically from sender and, then sender finally decides the MTR through EWMA (Exponential Weighted Moving Average) algorithm. Here, MTR has to be exactly estimated because available bandwidth is calculated with difference of MTR and current transfer rate. However, when network is congested due to traffic load and where competing flows are coexisted, it bring about a severe fairness problem. This paper proposes a congestion degree based MTR estimation algorithm. Here, the congestion degree stands a relative index for current congestion status on bottleneck link, which is calculated with arriving intervals of a pair packets. The algorithm try to more classify depending on the congestion degree to estimate more actual available bandwidth. With the network simulation results, our proposed method showed that the fairness problem among the competing flows is significantly resolved in comparison with that of UDT.

• Journal of Broadcast Engineering
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• v.24 no.1
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• pp.118-131
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• 2019

In order to provide 8K UHD contents of terrestrial broadcasting with a large capacity, the terrestrial broadcasting system has various problems such as limited bandwidth and so on. To solve these problems, UHD contents transmission technology has been actively studied, and an 8K UHD broadcasting system using terrestrial broadcasting network and communication network has been proposed. The proposed technique is to solve the limited bandwidth problem of terrestrial broadcasting network by segmenting 8K UHD contents and transmitting them to heterogeneous networks through hierarchical separation. Through the terrestrial broadcasting network, the base layer corresponding to FHD and the additional enhancement layer data for 4K UHD are transmitted, and the additional enhancement layer data corresponding to 8K UHD is transmitted through the communication network. When 8K UHD contents are provided in such a way, user can receive up to 4K UHD broadcasting by terrestrial channels, and also can receive up to 8K UHD additional communication networks. However, in order to transmit the 4K UHD contents within the allocated bit rate of the domestic terrestrial UHD broadcasting, the compression rate is increased, so a certain level of image deterioration occurs inevitably. Due to the nature of UHD contents, video quality should be considered as a top priority over other factors, so that video quality should be guaranteed even within a limited bit rate. This requires packet scheduling of content generators in the broadcasting system. Since the multiplexer sends out the packets received from the content generator in order, it is very important to make the transmission time and the transmission rate of the process from the content generator to the multiplexer constant and accurate. Therefore, we propose a variable transmission scheduler between the content generator and the multiplexer to guarantee the image quality of a certain level of UHD contents in this paper.

• Progress in Medical Physics
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• v.14 no.2
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• pp.124-130
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• 2003

Currently, as a consequence of PACS (Picture Archiving Communication System) implementation many hospitals are replacing conventional film-type interpretations of diagnostic medical images with new digital-format interpretations that can also be saved, and retrieve However, the big limitation in PACS is considered to be the lack of mobility. The purpose of this study is to determine the optimal communication packet size. This was done by considering the terms occurred in the wireless communication. After encoding medical image using JPGE2000 image compression method, This method embodied auto-error correction technique preventing the loss of packets occurred during wireless communication. A PC class server, with capabilities to load, collect data, save images, and connect with other network, was installed. Image data were compressed using JPEG2000 algorithm which supports the capability of high energy density and compression ratio, to communicate through a wireless network. Image data were also transmitted in block units coeded by JPEG2000 to prevent the loss of the packets in a wireless network. When JPGE2000 image data were decoded in a PUA (Personal Digital Assistant), it was instantaneous for a MR (Magnetic Resonance) head image of 256${\times}$256 pixels, while it took approximately 5 seconds to decode a CR (Computed Radiography) chest image of 800${\times}$790 pixels. In the transmission of the image data using a CDMA 1X module (Code-Division Multiple Access 1st Generation), 256 byte/sec was considered a stable transmission rate, but packets were lost in the intervals at the transmission rate of 1Kbyte/sec. However, even with a transmission rate above 1 Kbyte/sec, packets were not lost in wireless LAN. Current PACS are not compatible with wireless networks. because it does not have an interface between wired and wireless. Thus, the mobile JPEG2000 image viewing system was developed in order to complement mobility-a limitation in PACS. Moreover, the weak-connections of the wireless network was enhanced by re-transmitting image data within a limitations The results of this study are expected to play an interface role between the current wired-networks PACS and the mobile devices.

• Journal of Korea Multimedia Society
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• v.14 no.1
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• pp.54-75
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• 2011

PoC(Push-to-talk Over Cellular) is an integrated technology of group voice calls, video calls and internet based multimedia services. If a PoC user can not participate in the PoC session for various reasons such as an emergency situation, lack of battery capacity, then the user can use the PoC Box which has a similar functionality to the MM Box in the MMS(Multimedia Messaging Service). The RTSP(Real-Time Streaming Protocol) method is recommended to be used when there is a transmission session between the PoC box and a terminal. Since the existing VOD service uses a wired network, the packet size of RTSP-based VOD service is huge, however, the PoC service has wireless communication environments which have general characteristics to be used in RTSP method. Packet loss in a wired communication environments is relatively less than that in wireless communication environment, therefore, a buffering latency occurs in PoC service due to a play-out delay which means an asynchronous play of audio & video contents. Those problems make a user to be difficult to find the information they want when the media contents are played-out. In this paper, the following techniques and methods were proposed and their performance and superiority were verified through testing: cross-over dual reception buffering technique, advance partition multi-reception buffering technique, and on-demand multi-reception buffering technique, which are designed for effective picking up of information in media content being transmitted in short amount of time using RTSP when a user searches for media, as well as for reduction in playback delay; and same-priority packetization transmission method and priority-based packetization transmission method, which are media data packetization methods for transmission. From the simulation of functional evaluation, we could find that the proposed multiple receiving buffering and packetizing methods are superior, with respect to the media retrieval inclination, to the existing single receiving buffering method by 6-9 points from the viewpoint of effectiveness and excellence. Among them, especially, on-demand multiple receiving buffering technology with same-priority packetization transmission method is able to manage the media search inclination promptly to the requests of users by showing superiority of 3-24 points above compared to other combination methods. In addition, users could find the information they want much quickly since large amount of informations are received in a focused media retrieval period within a short time.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.4
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• pp.31-40
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• 2016

In vehicular ad-hoc networks (VANETs), vehicles sense information on emergency incidents (e.g., accidents, unexpected road conditions, etc.) and propagate this information to following vehicles and a server to share the information. However, this process of emergency message propagation is based on multiple broadcast messages and can lead to broadcast storms. To address this issue, in this work, we use a novel approach to detect the vehicles that are farthest away but within communication range of the transmitting vehicle. Specifically, we discuss a signal-to-noise ratio (SNR)-based linear back-off (SLB) scheme where vehicles implicitly detect their relative locations to the transmitter with respect to the SNR of the received packets. Once the relative locations are detected, nodes that are farther away will set a relatively shorter back-off to prioritize its forwarding process so that other vehicles can suppress their transmissions based on packet overhearing. We evaluate SLB using a realistic simulation environment which consists of a NS-3 VANET simulation environment, a software-based WiFi-IP gateway, and an ITS server operating on a separate machine. Comparisons with other broadcasting-based schemes indicate that SLB successfully propagates emergency messages with latencies and hop counts that is close to the experimental optimal while reducing the number of transmissions by as much as 1/20.

• Journal of KIISE
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• v.43 no.12
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• pp.1428-1436
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• 2016

A multipath routing in wireless sensor networks (WSNs) provides advantage such as reliability improvement and load balancing by transmitting data through divided paths. For these reasons, existing multipath routing protocols divide path appropriately or create independent paths efficiently. However, when the sink node moves to avoid hotspot problem or satisfy the requirement of the applications, the existing protocols have to reconstruct multipath or exploit foot-print chaining mechanism. As a result, the existing protocols will shorten the lifetime of a network due to excessive energy consumption, and lose the advantage of multipath routing due to the merging of paths. To solve this problem, we propose a multipath creation and maintenance scheme to support the mobile sink node. The proposed protocol can be used to construct local grid structure with restricted area and exploit grid structure for constructing the multipath. The grid structure can also be extended depending on the movement of the sink node. In addition, the multipath can be partially reconstructed to prevent merging paths. Simulation results show that the proposed protocol is superior to the existing protocols in terms of energy efficiency and packet delivery ratio.

• The Transactions of the Korea Information Processing Society
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• v.3 no.7
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• pp.1812-1821
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• 1996

In this paper, firstly we describe the structure and the performance of our ICPS(Information Communicaion Processing System) which currently provides information retrieval services, and then make a proposal for the construction of the open-networking information communication infra-structure which enables us to fully pre-pare for the emerging information society. In detail, the structure and the methodology needed for the implementation of the billing function on behalf of all information providers by using the user access network number as a user identification number while guaranteeing the equivalent access to the multiple value-added networks, are suggested. Based on the above ideas, the AICPS(Advanced Information Communication Processing System) has been designed and implemented. Final system performance evaluation with the assumption of a poling system as a system model, shows that our system can handle 10,000 user simultaneously who are using V.34 28.8 kbps modems and the processing capacity is 288,000 packet/sec. This result is so far superior to our target performance established during the desingning procedure. Namely, our system was originally designed to accommodate only 960 users at the same time. By taking advantage of this excessive high performance of our system, many other users can easily access the new services which are accessible only throught the ISDN or the Internet.