• Journal of the Institute of Electronics Engineers of Korea CI
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• v.45 no.5
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• pp.117-124
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• 2008

In this paper, the call setup performance of the CSCF(Call Session Control function) that manages the session control in providing the IMS(IP Multimedia subsystem)-based multimedia service in the BcN(Broadband convergence Network) is analyzed. While the performance related with the call/session of the SIP-Based voice service is analysed, the study for the call recessing performance of the IMS-based multimedia service is insufficient. In this paper, as, the processing capacity and subscriber number of CSCF were changed, the call setup delay time according to the session setup procedure of IMS was measured using the M/M/1 queuing model and OPNET simulation. The experimental results show that the sudden increased delay showed up in case the session establishment requirement ratio of total users over 13% of the CSCF processing capacity. Therefore, the user number and capacity of CSCF suitable for the session establishment delay threshold can be calculated or estimated.

• Journal of the Korean Society of Surveying, Geodesy, Photogrammetry and Cartography
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• v.20 no.1
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• pp.69-75
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• 2002

The important issues of facility database management in GIS are to collect up-to-date information and to update information in accordance with new-establishment, repair and replacement of the facilities. Therefore, it is necessary to develop a system which has capability of monitoring facilities as well as managing database efficiently. The purpose of this study is to propose possibility of implementation of mobile GIS solution for the facility management. In order to achieve the goal, to process existing digital maps and to receive on-site information through the wireless communication service are required. In addition, the system is required to process spatial information obtained by GPS and digital photogrammetric technique with real-time updating database in server. The system increases efficiency both in work flow and monitoring for facility management by providing optimal routing information to the sites and real-time two-way communication using VoIP(Voice over Internet Protocol). The system is expected to perform real-time database management effectively. In consequence, the system could appropriately response on-site situations in various practical applications. The proposed technology could contribute to improve nation's leading-edge technology.

• Journal of Intelligence and Information Systems
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• v.26 no.4
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• pp.87-109
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• 2020

Currently, thanks to the major stride made in developing wired and wireless communication technology, a variety of IT services are available on land. This trend is leading to an increasing demand for IT services to vessels on the water as well. And it is expected that the request for various IT services such as two-way digital data transmission, Web, APP, etc. is on the rise to the extent that they are available on land. However, while a high-speed information communication network is easily accessible on land because it is based upon a fixed infrastructure like an AP and a base station, it is not the case on the water. As a result, a radio communication network-based voice communication service is usually used at sea. To solve this problem, an additional frequency for digital data exchange was allocated, and a ship ad-hoc network (SANET) was proposed that can be utilized by using this frequency. Instead of satellite communication that costs a lot in installation and usage, SANET was developed to provide various IT services to ships based on IP in the sea. Connectivity between land base stations and ships is important in the SANET. To have this connection, a ship must be a member of the network with its IP address assigned. This paper proposes a SANET-CC protocol that allows ships to be assigned their own IP address. SANET-CC propagates several non-overlapping IP addresses through the entire network from land base stations to ships in the form of the tree. Ships allocate their own IP addresses through the exchange of simple requests and response messages with land base stations or M-ships that can allocate IP addresses. Therefore, SANET-CC can eliminate the IP collision prevention (Duplicate Address Detection) process and the process of network separation or integration caused by the movement of the ship. Various simulations were performed to verify the applicability of this protocol to SANET. The outcome of such simulations shows us the following. First, using SANET-CC, about 91% of the ships in the network were able to receive IP addresses under any circumstances. It is 6% higher than the existing studies. And it suggests that if variables are adjusted to each port's environment, it may show further improved results. Second, this work shows us that it takes all vessels an average of 10 seconds to receive IP addresses regardless of conditions. It represents a 50% decrease in time compared to the average of 20 seconds in the previous study. Also Besides, taking it into account that when existing studies were on 50 to 200 vessels, this study on 100 to 400 vessels, the efficiency can be much higher. Third, existing studies have not been able to derive optimal values according to variables. This is because it does not have a consistent pattern depending on the variable. This means that optimal variables values cannot be set for each port under diverse environments. This paper, however, shows us that the result values from the variables exhibit a consistent pattern. This is significant in that it can be applied to each port by adjusting the variable values. It was also confirmed that regardless of the number of ships, the IP allocation ratio was the most efficient at about 96 percent if the waiting time after the IP request was 75ms, and that the tree structure could maintain a stable network configuration when the number of IPs was over 30000. Fourth, this study can be used to design a network for supporting intelligent maritime control systems and services offshore, instead of satellite communication. And if LTE-M is set up, it is possible to use it for various intelligent services.

• Journal of Korea Multimedia Society
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• v.17 no.12
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• pp.1453-1460
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• 2014

WFQ (Weighted Fair Queuing) provides not only fairness among traffic flows in using bandwidth but also guarantees the Quality of Service (QoS) that individual flow requires, which is why it has been applied to the resource reservation protocol (RSVP)-capable router. The RSVP allocates an enough resource to satisfy both the rate and end-to-end delay requirements of the flow in the condition of no packet loss, and the WFQ scheduler guarantees those QoS requirements with the allocated resource. In practice, however, most QoS-guaranteed services allow a degree of packet loss, especially from 0.1% to 3% for Voice over IP. This paper discovers that the packet loss rate of each traffic flow is determined by only its time-stamp adjustment value, and then enhances the WFQ to provide a differentiated packet loss guarantee under general traffic conditions in terms of both traffic characteristics and QoS requirements. The performance evaluation showed that the proposed WFQ could increase the utilization of bandwidth by 8~11%.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.5 s.347
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• pp.7-16
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• 2006

IPTV is referred to the service which provide integrated IPTV services for providing video, 10/100-Mbit/sec Internet, voice, video-on-demand (VOD), and other broadband applications including home security, video conferencing, and telemedicine. All services are integrated into an IP (Internet Protocol) architecture designed specifically for Gigabit Ethernet FTTH systems, HFC or xDLC. It is absolutely necessary that telecon operators provide IP video delivery platforms that enable service providers to transform their business. With their own products, they can better manage their existing services and generate new revenues from broadcast TV, movies on demand and multimedia. Triple-play is a combination of broadcast, telephony and broadband services offered through IPTV networks. With cable operators allowed to offer a triple-play bundle, the nation's telecom operators are beginning to get a little anxious. Cable operators assert that triple-play is a must-have and natural extension of the cable service bundle. The Korean Cable TV Association asserts that the triple-play model is of paramount importance to the cable industry's future growth. But the telecom sector considers itself unfairly disadvantaged, saying they cannot compete until regulatory issues are resolved. The start of web-based television in Korea may still be some time off with a confrontation between the nation's IT regulator and broadcasting sector over the service's legal boundaries shows no signs of being resolved my time soon. korea should be is the fastest-growing provider of IPTV solutions in the industry, with over worldwide customers.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.14 no.4
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• pp.247-256
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• 2021

In this paper, we propose a packet loss concealment (PLC) algorithm using pitch harmonic motion prediction and adaptive signal amplitude prediction and. The spectral motion prediction method divides the spectral motion of the previous usable frame into predetermined sub-bands to predict and restore the motion of the lost signal. In the proposed algorithm, the speech signal is classified into voiced and unvoiced sounds. In the case of voiced sounds, it is further divided into pitch harmonics using the pitch frequency to predict and restore the pitch harmonic motion of the lost frame, and for the unvoiced sound, the lost frame is restored using the spectral motion prediction method. When the continuous loss of speech frames occurs, a method of adjusting the gain using the least mean square (LMS) predictor is proposed. The performance of the proposed algorithm was evaluated through the objective evaluation method, PESQ (Perceptual Evaluation of Speech Quality) and was showed MOS 0.1 improvement over the conventional method.

• Proceedings of the Korea Society for Simulation Conference
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• 2001.10a
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• pp.440-443
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• 2001

As inter-network traffics grows rapidly, the router systems as a network component becomes to be capable of not only wire-speed packet processing but also plentiful programmability for quality services. A network processor technology is widely used to achieve such capabilities in the high-end router. Although providing two such capabilities, the network processor can't support a deep packet processing at nominal wire-speed. Considering QoS may result in performance degradation of processing packet. In order to achieve foster processing, one chipset of network processor is occasionally not enough. Using more than one urges to consider a problem that is, for instance, an out-of-order delivery of packets. This problem can be serious in some applications such as voice over IP and video services, which assume that packets arrive in order. It is required to develop an effective packet processing mechanism leer using more than one network processors in parallel in one linecard unit of the router system. Simulation analysis is also needed for verifying the mechanism. We propose the packet processing mechanism consisting of more than two NPs in parallel. In this mechanism, we use a load-balancing algorithm that distributes the packet traffic load evenly and keeps the sequence, and then verify the algorithm with simulation analysis. As a simulation tool, we use DEVSim++, which is a DEVS formalism-based hierarchical discrete-event simulation environment developed by KAIST. In this paper, we are going to show not only applicability of the DEVS formalism to hardware modeling and simulation but also predictability of performance of the load balancer when implemented with FPGA.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.3
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• pp.569-575
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• 2014

These days as smart TVs or android set top boxes have come into wide use, VoIP(Voice over Internet Protocol) services such as the Skype which can beused on computer system, can be also accessed by smart TVs or set top boxes. and the various kinds of distance learning programs have been developed on smart TVs or set top boxes. but when users have utilized a webcam with smart TVs or set top boxes, users have to go in front of smart TVs or set top boxes In this research paper, the new technology can be solved the problem of webcam use according to chaned environment. This tech is related with the wireless webcam one which can be applicable to open spaces such as living room with some distance between user and smart TV. There are few problems in long distance the test results show high reliability in living room with some distance.

• International Journal of Computer Science & Network Security
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• v.23 no.4
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• pp.172-178
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• 2023

When the mobile device moves from the coverage of one access point to the radio coverage of another access point it needs to maintain its connection with the current access point before it successfully discovers the new access point, this process is known as handoff. During handoff the acceptable delay a voice over IP application can bear is of 50ms whereas the delay on medium access control layer is high enough that goes up to 350-500ms. This research provides a suitable methodology on medium access control layer of the IEEE 802.11 network. The medium access control layer comprises of three phases, namely discovery, reauthentication and re-association. The discovery phase on medium access control layer takes up to 90% of the total handoff latency. The objective is to effectively reduce the delay for discovery phase to ensure a seamless handoff. The research proposes a scheme that reduces the handoff latency effectively by scanning channels prior to the actual handoff process starts and scans only the neighboring access points. Further, the proposed scheme enables the mobile device to scan first the channel on which it is currently operating so that the mobile device has to perform minimum number of channel switches. The results show that the mobile device finds out the new potential access point prior to the handoff execution hence the delay during discovery of a new access point is minimized effectively.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.3
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• pp.997-1005
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• 2010

Recently, there has been a growing interest in the use of WLAN technology due to its easy deployment, flexibility and so on. Examples of WLAN applications range from standard internet services such as Web access to real-time services with strict latency/throughput requirements such as multimedia video and voice over IP on wireless network environments. Fair and efficient distribution of the traffic loads among APs(Access Points) has become an important issue for improved utilization of WLAN. This paper focuses on an AP selection scheme for achieving better load balance, and hence increasing network resource utilization for each user on wireless network environments. This scheme makes use of active scan patterns and the network delay as main parameters of load measurement and AP selection. This scheme attempts to estimate the AP traffic loads by observing the up/down delay and utilize the results to maximize the link resource efficiency through load balancing. We compared the proposed scheme with the original SNR(Signal to Noise Ratio)-based scheme using the NS-2(Network Simulation.2). We found that the proposed scheme improves the throughput by 12.5% and lower the network up/down link delay by 36.84% and 60.42%, respectively. All in all, the new scheme can significantly increase overall network throughput and reduce up/down delay while providing excellent quality for voice and video services.