• Journal of KIISE:Information Networking
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• v.30 no.2
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• pp.260-269
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• 2003

The traditional distributed object middleware platform is not suitable for the transmission of stream data, because RPC(Remote Procedure Call)-based message transmission have a great overhead. Therefore, the OMG(Object Management Group) proposes the AV(Audio and Video) stream reference model for streaming on the distributed object middleware platform. But, this reference model has not a detail of implementation. Particularly it also has not congestion control scheme for improvement of network efficiency on the real network environment. It is a very important and difficult technical issue to provide the stream transmission platform with advanced congestion control scheme. In this paper, we propose an architecture of a distributed stream transport platform and deal with the design and implementation concept of our proposed platform. Also, we present a mechanism to improve streaming utilization by SRTP(Smart RTP). SRTP is our proposed TCP-Friendly scheme.

• Journal of KIISE:Information Networking
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• v.29 no.5
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• pp.501-509
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• 2002

When using TCP over a mobile network, TCP responds to a handoff by invoking a congestion control algorithm, thereby resulting in a degraded end-to-end performance in a mobile network. In this paper, two schemes are proposed, TCP-MD and TCP-R. TCP-MD can detect the movement of s mobile host early on, whereas TCP-R can force the source to freeze data transmission during registration. The proposed schemes maintain end-to-end TCP semantics, making it possible to fully interoperate with the existing infrastructure. Only a small change is required in the mobile host, plus the implementation is simple because some Mobile IP messages are used to notify the handoff, eliminating the need for any additional messages. Simulations confirmed that the proposed schemes give an excellent performance under various environments.

• Journal of KIISE:Information Networking
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• v.27 no.1
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• pp.76-87
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• 2000

ATM is a key technology of integration of multimedia service. Recently, Many study have been concentrated on performance testing for evaluation network performance are stronger everyday. The performance testing is on evaluation of maximal throughput of network by measuring and analyzing of various performance parameters. There are two ways to test ATM network performance; one is using QoS in cell level on the point of network's view, and the other is using metric in frame level in the point of user's view. And, the standardization process is also under way. In this paper, we derive a performance requirement of TCP in TCP/IP data transmission over ATM UBR service. By applying the derived requirements to ATM and packet networks, we evaluate the performance of TCP over UBR based on the result of our simulations. Therefore, we evaluate the result of simulation and find degradation of network throughput by interaction between TCP congestion control and ATM cell drop policy. So we suggest the accelerated Vegas that modify traditional TCP Vegas in congestion control mechanism for batter network throughput.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.3
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• pp.48-55
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• 2015

This paper studies how to solve a problem caused by M2M terminals sending a few data based on $3^{rd}$ Generation Partnership Project(3GPP) Long Term Evolution-Advanced(LTE-A) system and then it is analyzed, proposed, and introduced into the techniques. Especially, it is introduced solution for the lack of Random Access Channel and an increasing number of latency caused by countless M2M devices. It is proposed the technology for M2M grouping as well as allowable access probability according to user type. As it decreases the number of terminal by grouping M2M devices to try random access at PRACH, it can be reduced collision between Cellular users and M2M devices. So, it is proved that the proposed mechanism can solve the increasing average latency of random access on system coexisting Cellular users and M2M devices through simulations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.5
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• pp.708-720
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• 1993

The new preventive control mechanisms for traffic management in BISDN/ATM networks can be divided into Connection Admission Control(CAC), Usage Parameter Control (UPC), and Priority Control. Of these mechanism, Usage Parameter Control continuously monitors the parameters admitted in the network's entry point to guarantee quality of service of connections already admitted. Upon detecting traffic that violates the negotiated parameter, it takes the necessary control measures to prevent congestion. Among these traffic control methods, this paper focuses on the Usage Parameter Control method, and proposes and designs GRACE-LB(Guaranteed Rate Acceptance & Control Element-using Leaky Bucket) which improves upon existing UPC models. GRACE-LB modifies the previous LB model by eliminating the cell buffer, dividing the token Pool into two pools, Long-term pool, Short-term pool, and changing the long-term token generating form using 'Cycle Token' into the same bursty form as the traffic source. Through this, GRACE-LB achieves effective control of the Average Bit Rate(ABR) and burst duration of bursty multimedia traffic which previous LB models found difficult to control. Also, since GRACE-LB can e implemented using only simple operations and there are no cell buffers in it, it has the merit of being easily installed at any place.

• Journal of Advanced Navigation Technology
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• v.15 no.3
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• pp.355-363
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• 2011

Transcoding in the wireless mobile network is an important mechanism that reduces the delay time and improves the stream processing capacity. Wireless mobile streaming media services, however, have such problems as congestion, interference and delay due to narrow network bandwidth and limited resources. These problems degrade not only Quality of Service (QoS) but also responsiveness of the streaming media service. To solve this problem, this paper proposes a object version transcoding method. The proposed method analyzes the object versions to construct the transcoding graph. This paper utilizes a reference rate-based control function for an efficient streaming, and measures MVDS(Multiple Version Delay Saving) for an efficient delay savings. The simulation results show that the proposed mechanism achieves improved performance in delay rate and cache hit rate compared with those of other existing methods.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.349-360
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• 2002

In ITU H.323 teleconference system, the RTP/RTCP protocol is offered to transfer real-time multimedia stream. Both sender and receiver hate experience in packet loss and jitter which result from network congestion over Internet. Audio quality over Internet depends on the number of lost packets and on jitter between successive packets. The goal of our study is to improve the speech quality over Internet by checking the packet loss characteristics of the network and adopting the but for control management mechanism at the receiver. We suggest a dynamic redundant audio transmission mechanism which examines the packet loss rate and uses the feedback information through RTCP.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.3 no.6
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• pp.575-596
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• 2009

Facing limited network resources such as bandwidth and processing capability, the Internet will have congestion from time to time. In this paper, we propose a scheme to maximize the total utility offered by the network to the end user during congested times. We believe the only way to achieve our goal is to make the scheme application-aware, that is, to take advantage of the characteristics of the application. To make our scheme scalable, it is designed to be class-based. Traffic from applications with similar characteristics is classified into the same class. We adopted the RED queue management mechanism to adaptively control the traffic belonging to the same class. To achieve the optimal utility, the traffic belonging to different classes should be controlled differently. By adjusting link bandwidth assignments of different classes, the scheme can achieve the goal and adapt to the changes of dynamical incoming traffic. We use the control theoretical approach to analyze our scheme. In this paper, we focus on optimizing the control on two types of traffic flows: TCP and Simple UDP (SUDP, modeling audio or video applications based on UDP). We derive the differential equations to model the dynamics of SUDP traffic flows and drive stability conditions for the system with both SUDP and TCP traffic flows. In our study, we also find analytical results on the TCP traffic stable point are not accurate, so we derived new formulas on the TCP traffic stable point. We verified the proposed scheme with extensive NS2 simulations.

• Journal of Communications and Networks
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• v.16 no.3
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• pp.344-354
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• 2014

Transmission control protocol friendly rate control (TFRC) is designed to mainly provide optimal service for unicast applications, such as multimedia streaming in the best-effort Internet environment. However, high bandwidth networks with large delays present an environment where TFRC may have a problem in utilizing the full bandwidth. TFRC inherits the slow-start mechanism of TCP Reno, but this is a time-consuming process that may require many round-trip-times (RTTs), until an appropriate sending rate is reached. Another disadvantage inherited from TCP Reno is the RTT-unfairness problem, which severely affects the performance of long-RTT flows. In this paper, we suggest enhanced TFRC for high quality video streaming over high bandwidth delay product networks. First, we propose a fast startup scheme that increases the data rate more aggressively than the slow-start, while mitigating the overshooting problem. Second, we propose a bandwidth estimation method to achieve more equitable bandwidth allocations among streaming flows that compete for the same narrow link with different RTTs. Finally, we improve the responsiveness of TFRC in the presence of severe congestion. Simulation results have shown that our proposal can achieve a fast startup and provide fairness with competing flows compared to the original TFRC.

• Proceedings of the Korean Information Science Society Conference
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• 2005.11a
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• pp.154-156
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• 2005

본 논문은 TCP의 무선 환경에서의 전송 성능 감소와 패킷 왕복 시간에 따른 대역폭 선점 문제를 해결하기 위한 타이머 기반의 혼잡 제어 방식을 제안한다. TCP는 패킷 손실 확률이 많은 무선 환경에서 네트워크 혼잡에 의한 패킷 손실을 방지하기 위한 느린 전송률 증가로 인해 전송 성능이 크게 떨어진다. 또한, TCP 송신자는 전송률을 결정하는 전송 윈도우를 수신자로부터 응답 메시지를 받을 때만 조정시키므로, 패킷의 왕복 시간의 차이에 따른 전송률 편중 현상과 다수개의 응답 메시지에 의한 과도한 트래픽 발생의 문제가 발생한다. 본 논문에서 제안하는 타이머 기반의 TCP 혼잡 제어 방식은 패킷의 전송 시간 간격을 타이머로 조정함으로써 무선 환경에서 전송 성능을 향상시키고 패킷의 왕복 시간 차이에 따른 전송률 편중 현상을 완화시키며 다수개의 응답 메시지에 의한 과도한 트래픽의 발생을 제한한다. 제안하는 방법은 실제 환경에서 구현되었으며, 다양한 네트워크 환경에서의 실험을 통해 그 성능을 입증하였다.