• Journal of the Institute of Convergence Signal Processing
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• v.17 no.1
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• pp.46-52
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• 2016

LTE-Advanced system consisting of the number of cells in the cellular environment because it is built to allow efficient use of limited frequency resources of adjacent cell interference avoidance should be considered. Transition services in accordance with the development of the mobile communication technology, wireless multimedia content from voice-centric mobile communications services and causing a lot of mobile data traffic, such as smart phones and tablet terminals spread of a data-driven surge in mobile data traffic base stations in urban areas by increasing became a reality that can not be prevented. In this paper, we propose a new Hybrid resource allocation technique for improving the performance of the cell boundary and analyzed the performance of the proposed new techniques to perform the simulation using LTE-Advanced system level simulator based on 19cell of cellular system model.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.3
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• pp.111-119
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• 2010

With the rapid progress of wireless technologies today, mobile terminals with multiple access interfaces are emerging. In recent years, WLAN (Wireless Local Area Networks) has become the premier choice for many homes and enterprises. WiMAX (Worldwide Interoperability for Microwave Access) has also emerged as the wireless standard that aims to deliver data over long distances. Therefore, it is important to explore efficient integration methods for delivering multimedia data between heterogeneous wireless networks. In this paper, we developed the simulation models and environments for the mobile multihomed node that has both WiMAX and WLAN interfaces and can move around in both networks by using mobile IP. In order to verify the developed models, we designed and constructed several simulation scenarios, e.g. movement in WiMAX/WLAN, group mobility, MANET, and nested MIP under the various traffic environments such as oneway or bothway UDP packets, FTP traffic, and voice with SIP protocol. The simulation results show that the developed models are useful for mobility studies in various integrated wireless networks.

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.6
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• pp.689-694
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• 2014

DDoS attacks on MANET makes disable any node which support network function, cause critical results as a stopping of entire network service or separation to some parts. Packet aggregation, which gather some pieces of short length into a certain length of data, improves transmission performances on MANETs. In this paper, some effects on transmission performance of packet aggregation transmission be caused by DDoS Attacks are measured and analyzed in point of end-to-end level. MANET simulator, based on NS-2, is used for measurement end-to-end performances. MOS, connection rate, delay and packet loss rate is used as performance parameters. VoIP traffic is used for object service measurement. Finally, it is suggested that number of packet aggregation is more then 4 for keeping the transmission quality over MANETs under DDoS attacks.

• The KIPS Transactions:PartC
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• v.9C no.2
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• pp.267-276
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• 2002

By using wireless terminal, User that want to transmit multimedia traffic as well as simple text and voice have a tendency to increase. This paper proposes a fast handoff protocol that is suitable transmission for real-tim of multimedia traffic by using modified local registration. The proposed protocol solves the Triangle Routing Protocol that is one of existing opened issues without modifying protocol of CN, and we propose the method that MN is able to received a packet by real-time, even if MN is being handoff. We compare fast handoff protocol proposed in this paper with existing method n the registration cost and data packet transmission cost. As a result, we showed that fast handoff protocol proposed in this paper outperforms existing method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6A
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• pp.458-467
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• 2009

Long Tenn Evolution (LTE) is the next generation mobile phone technology which has being standardized by the Third Generation Partnership Project (3GPP). In the existing mobile communication networks, voice traffic is delivered through circuit switched networks. In LTE, however, all kinds of traffic are transferred through IP based packet switched networks which has best-effort characteristic. Therefore, providing QoS in LTE system is difficult. In order to provide QoS in LTE, RRM is very important. Especially, in part of RRM, call admission control (CAC) performs an important function to reduce network congestion and guarantee a certain level of QoS for on-going calls. In this paper, we propose a CAC algorithm in order to provide QoS for various kinds of services in LTE system. The performance of the proposed algorithm is evaluated with various simulation environments. The results show that the proposed algorithm provides QoS through rejections of requested calls. Especially, the proposed CAC algorithm can be satisfied with packet delay requirement defined in LTE specification.

• 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.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.15 no.4
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• pp.387-396
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• 2004

A scheme to evaluate the number of users and cell coverage of a WCDMA supporting multi-rate traffic is newly presented through calculation of the realizable Erlang capacity from a derived blocking probability and the path loss from the COST231 Walfisch-Ikegami(W) model. We evaluate the voice-data Erlang capacities at various data rates of 15 kbps to 960 kbps and it is shown that they have a linear relationship to each other. When the E$\_$b//N$\_$o/ is low from 4 ㏈ to 3 ㏈ in case of voice capacity of 50 Erlang at 8 kbps, the result shows the increase for the data capacity of 10 Erlang and the enlargement of 100 m for the cell coverage at low rate of 15 kbps, and the increase of 0.11 Erlang and the enlargement of 40 m at high rate of 960 kbps. The increase of the blocking probability results in the increase of the Erlang capacity, but not an effect on the cell coverage, and the increase of active users in a cell results in the decrease of the coverage.

• Convergence Security Journal
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• v.12 no.1
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• pp.35-42
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• 2012

Increase of data traffic and the advent of new real-time services require to change from the traditional TDM-based (Time Division Multiplexing) networks to the optical networks that soft and dynamic configuration. Voice and lease line services are main service area of the traditional TDM-based networks. This optical network became main infrastructure that offer many channel that can convey data, video, and voice. To provide high resilience against failures, Packet-optical networks must have an ability to maintain an acceptable level of service during network failures. Fast and resource optimized lightpath restoration strategies are urgent requirements for the near future Packet-optical networks with a Generalized Multi-Protocol Label Switching(GMPLS) control plane. The goal of this paper is to provide packet-optical network with a hierarchical multi-layer recovery in order to fast and coordinated restoration in packet-optical network/GMPLS, focusing on new implementation information. The proposed schemes do not need an extension of optical network signaling (routing) protocols for support.

• The KIPS Transactions:PartA
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• v.9A no.3
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• pp.317-326
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• 2002

The mobile telecommunication system has been growing exponentially after 1990s due to the high population in a city and the growth of mobile user. In this time, the current mobile system mainly concentrates on the voice communication. However, in the next generation, mobile users want to get very diverse services via mobile terminal such as the Internet access, web access, multimedia communication, and etc. For this reason, the next generation system, such as the UMTS system, has to support the packet data service and it will play the major role in the system. In different from the voice service, the packet data service should store packet data in a buffer at base station before it forwards. Consequently, the performance closely related to the buffer management and its operation. However, until now, there are rare study on the buffer behavior and management. In this paper, we will observe the buffer behavior in the UMTS systems by using simulation and analyze the results. For this research, we generate packet data traffics and model the UMTS system for a simulation. Then we analyze the buffer state by a simulation and calculate the buffer overflow probability by mathematical methods.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.42 no.2 s.332
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• pp.89-96
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• 2005

In this paper, we propose a customer premise platform for processing multimedia data service on the ATM network. The proposed platform has a specific AAL2 processor that includes AAL2 protocol and scheduler algorithm so as to off-load large potion of burden from host processor and make it easy to process multimedia data from the ATM network in real time compared with conventional platform in which AAL/ATM tasks are processed by software. The ATS scheduler that is implemented based on 2-level time slot ring provides a simple and efficient method for scheduling data of VBR-rt, UBR and CBR traffics. TMS320C5402 DSP is used to process voice-related tasks such as voice compression and voice packet manupulation and AAL2 processor is implemented on $0.35\;{\mu}m$ process line. We implemented the customer premise equipment for VoDSL service and tested the proposed platform on a test bed network. The experimental results show that the proposed equipment has the call success rate of $97\%$ at least and provides voice service of toll-qualify.