• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.4B
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• pp.371-377
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• 2011

Users of the Information Age, IT usage patterns of the wired broadband information services and various forms of the same quality wireless multimedia services are required. Changes of these times the next-generation mobile communications (IMT-Advanced) has emerged as the necessity of developing its current voice and packet data communications on the move in the high-speed 100Mbps, 1Gbps in stationary and slow data transmission rates up to fixed-mobile convergence based on needed to provide ubiquitous service platform for the realization of IMT-Advanced is the time for preparation. In particular, 3-5GHz band, focused on mobile communications can be used to secure the necessary frequency band relocated and the existing crosstalk analysis methodology developed for the services rendered, and the frequency of such results to obtain new spectrum for IMT-Advanced for the country to secure the frequency characteristics and IMT-Advanced 3-5GHz band for the radio frequency of the characterization techniques necessary to develop a national wireless communication interference and frequency-based technology acquisition and management skills were identified.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.5A
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• pp.479-486
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• 2008

In this paper, an efficient training symbol design with m-sequence is proposed for the MIMO-OFDM based next generation wireless transmission system which supports gigabits per second data rate. In the traditional blute force method, the preamble design is based on the case by case comparison with the system requirements. This paper discusses a training symbol design methodology for the MIMO-OFDM system based on the m-sequence which has been widely used in the spread spectrum communication areas due to its good correlation characteristics. Also the step-by-step design and performance verification method within the limited search space is discussed. The proposed method targets the design of the training symbol which satisfies system requirements for the packet based MIMO-OFDM wireless communication system including automatic gain control(AGC), timing synchronization, frequency and sampling offset estimation, and MIMO channel estimation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.2
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• pp.296-304
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• 2010

MoIP is technology to transmit a variety of multimedia over IP, but compared to traditional voice services require greater bandwidth and radio resources in a wireless environment has already reached the limits. Therefore, as a way to resolve this issue for header compression is a lot of research. SCTP protocol header compression using ROHC-SCTP has been research, ROHC-SCTP packet structure of the ROHC algorithm with different types and, SCTP header compression to apply the characteristics of the poor performance of many of these have drawbacks. Therefore, in this paper to solve these problems better header compression algorithm was designed. In this paper, the proposed algorithm to evaluate the NS-2 simulation environment was modeled on the header compression operation. Evaluation results, the algorithm designed in this paper compared to ROHC-SCTP algorithms determine the overhead rate was low, the data types vary a lot better when the total header size was small.

• Journal of Communications and Networks
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• v.5 no.1
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• pp.73-81
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• 2003

The controlled load service defined within the IETF's Integrated Services architecture for quality-of-service (QoS) in the Internet requires source nodes to regulate their traffic while the network, in combination with an admission control strategy, provides a guarantee of performance equivalent to that achieved in a lightly loaded network. Packets sent in violation of the traffic contract are marked so that the network may assign them a lower priority in the use of bandwidth and buffer resources. In this paper, we define the requirements of a scheduler serving packets belonging to the controlled load service and present a novel scheduler that exactly achieves these requirements. In this set of requirements, besides efficiency and throughput, we include an additional important requirement to bound the additional delay of unmarked packets caused due to the transmission of marked packets while dropping as few marked packets as possible. Without such a bound, unmarked packets that are in compliance with the traffic contract are not likely to experience delays consistent with that in al lightly loaded network. For any given desired bound ${\alpha}$ on this additional delay, we present the CL(${\alpha}$) scheduler which achieves the bound while also achieving a per-packet work complexity of O(1) with respect to the number of flows. We provide an analytical proof of these properties of the CL(${\alpha}$) scheduler, and we also verify this with simulation using real traces of video traffic. The scheduler presented here may be readily adapted for use in scheduling flows with multi-level priorities such as in some real-time video streams, as well as in other emerging service models of the Internet that mark packets to identify drop precedences.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.8B
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• pp.1426-1437
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• 1999

The introduction of fast packet switching networks such as the ATM leads to the possibility of video services with constant quality. However, the fixed traffic descriptor(TD), which is defined by ATM Forum and ITU-T, can not simultaneously satisfy network efficiency and video quality. This paper presents fast resource reallocation protocol (FRRP) which can provide an effective compromise between network efficiency and video quality. This protocol reallocates the network resources and then adapts the traffic to the reallocated resources. In the proposed method, a video session is divided into several time-intervals according to its generated traffic and TD is calculated which well characterizes the traffic during the specific subdivided interval. In addition, a system model, where MPEG coding algorithm and network control are unified, is proposal and the performance of the FRRP in terms of cell loss rate(CLR) is evaluated though the simulation using real video traffic(Star Wars). Simulation results show that the FRRP has only 5 ~35% CLR of the fixed TD.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.9B
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• pp.793-805
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• 2003

In OBS (Optical Burst Switching) networks which decouple the burst from its header, the fault of a fiber link can lead to the failure of all the light-path that traverses the fiber. Because each light-path is expected to operate at a rate of a few Gbps by using WDM (Wavelength Division Multiplexing) technology, any failure may lead to large data loss. Therefore, an efficient recovery scheme must be provided. In this paper, we analyze network utilization and BCP (Burst Control Packet) loss rate according to each link failure by applying the conventional restoration schemes in OBS networks. And through these simulation results, an ASPR scheme is proposed improve the fault management scheme in terms of recovery time and throughput. Finally, We compare the performance of our proposed scheme with that of the conventional one with respect to burst loss rate, resource utilization and throughput by OPNET simulations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.3
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• pp.506-515
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• 2015

In this paper, we propose a group playback synchronization method using adaptive synchronization delay time by the bit rate of media to synchronize a play position of streaming media between mobile smart devices. This method consists of streaming server-side and client-side synchronization algorithms based on synchronization delay time which includes connection time, control packet transmission time, streaming data buffering time, and synchronization processing time. We implement the Android media player application with synchronization support using the proposed algorithms and present the result of performance evaluation.

• The KIPS Transactions:PartC
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• v.12C no.3 s.99
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• pp.449-456
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• 2005

Optical burst switching (OBS) overcomes the inefficient resource usage of optical circuit switching and minimizes the optical buffering requirement of optical packet switching. General switch management protocol (GSMP) is an open interface between a label switch and a controller, and it provides connection, configuration, performance, event management and synchronization. GSMP open interface in the OBS network allows the implementation of OBS switch to be simple by separating the data forward plane from the control plane. We defined managed objects to support connection, configuration, performance, and fault management for the management of OBS network in the GSMP open interface. We proposed the network management model, in which the above managed objects are distributed in a controller and an OBS switch according to network management functions. We verified the possibility of connection management using distributed network management model in the GSMP open interface of OBS network by implementing GSMP and network management functions with managed objects of OBS.

• The KIPS Transactions:PartC
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• v.9C no.1
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• pp.65-72
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• 2002

In this paper, we implement a high-speed swatch tilth the function for label integration to enhance the expansion of networks using the label space of routers efficiently on MPLS over ATM networks. We propose an appropriate hardware structure to support the VC-merge function and differentiated services simultaneously. In this paper, we use the adaptive congestion control method such as EPD algorithm in carte that there is a possibility of network congestion in output buffers of each core LSR. In addition, we justify the validity of the proposed VC-merge method through simulation and comparison to conventional Non VC-merge methods. The proposed VC-merge capable switch is modeled in VHDL. synthesized, and fabricated using the SAMSUNG 0.5um SOG process.

• Journal of KIISE:Computing Practices and Letters
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• v.8 no.3
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• pp.328-335
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• 2002

When we want to transfer video streaming data through computer networks, we will have to be allocated a larger bandwidth compared to a general application. In general, it causes a serious network overload inevitably due to the limited bandwidth. In this paper, in order to resolve the problem, we haute taken a method for transmitting video streaming data using RTP and RTCP. With RR(Receiver Report) packet in RTCP we will test it to check whether the traffic in a network has occurred or not. If it happened, we haute tried to reduce the overall network traffic by dynamically changing the quantization factor of the Motion JPEG that is one of the encoding styles in JMF. When the ratio of the average of transmission for each session to the average of overall transmission is greater than 7%, we should adjust the amount of data to be transmitted for each session to reach the session mean values. The experimental results show that the proposed method taken here reduces the overload effectively and therefore improves the efficiency for transmitting video streaming data.