• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.6
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• pp.707-713
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• 2016

In the IEEE 802.11 Wireless LAN environment, the common Access Point (AP) selection of the existing terminal is based on signal strength. However, the signal strength-based AP selection method does not ensure an optimal data rate. Recently, several AP selection methods to solve this problem have been suggested. However, when we select AP, these have a latency problem and don't consider dense environments of AP. In this paper, we confirm the problem of the conventional AP selection about the signal strength and the throughput through the actual measurement, and propose algorithm that selects AP by scoring link speed and wireless round trip time to compensate the problem. Furthermore, the proposed AP selection algorithm through the actual experiment proves the improved performance as compared with the existing methods.

• Journal of Auto-vehicle Safety Association
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• v.14 no.3
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• pp.48-59
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• 2022

A V2X system can be a candidate as a means to increase the stability of autonomous vehicles. In particular, in order to implement a Level 4 or higher autonomous driving system, the application of the V2X system is essential. Wireless communication technologies applicable to the V2X system include WAVE and C-V2X. Currently, the V2X service most used by autonomous driving systems is a service that provides signal phase and timing information and since real-time characteristic is a very important, verification of this service must be done. In this paper, we measured the time delay characteristics for providing signal phase and timing information using WAVE and LTE communication, and proposed a TOD-based signal phase and timing information generation method without using V2X communication system. To analyze the time delay characteristics, RTT (Round Trip Time) was measured as a result of the measurement. Average RTT using WAVE communication was 5.84ms and was 104.15ms with LTE communication. As a result of measuring the error between the signal phase and timing information generated based on TOD and the actual traffic light state, it was measured to be -0.284~3.784sec.

• The Journal of the Korea Contents Association
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• v.9 no.8
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• pp.73-83
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• 2009

TCP(Transmission Control Protocol) is currently used connection-oriented protocol as a typical transport layer protocol in the Internet. However, it has deficiency not be able to communicate with other TCP entities when any link included in the path is down because of single-homing on single path. SCTP(Stream Control Transmission Protocol) suggested as the new transport layer protocol supports multi-homing feature, which provides several paths between source and destination. It can communicate with other SCTP entities using alternate path even when any link on the primary path is down. This paper aims to measure and analyze the multi-homing effect of SCTP over TCP in case of link-down using NS-2 simulator. We classify SCTP into $SCTP_{single-homing}$ and $SCTP_{multi-homing}$ because SCTP with single-homing can also be used like TCP. We measured throughput and bandwidth utilization varying link-down duration, bandwidth, and RTT(round trip time), Simulation results show that throughput of $SCTP_{multi-homing}$ is more than that of TCP by 18 % on average. It is also shown that $SCTP_{multi-homing}$ on varying RTT and bandwidth increases the throughput of TCP by 'l7% and 9% on average, respectively in the link-down environment. In above cases, more or less difference between $SCTP_{single-homing}$ and TCP on throughput and bandwidth utilization was found To summarize, multi-homing effect of SCTP over TCP on throughput is about 18 % on average in the link-down environment This experimental result can be used as the benchmark in order to estimate the multi-homing effect of SCTP over TCP when the link-down happens in the real Internet.

• Journal of KIISE:Information Networking
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• v.31 no.3
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• pp.269-279
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• 2004

The most important factor influencing the robustness of the Internet Is the end-to-end TCP congestion control. However, the congestion control scheme of TCP Reno, the most popular TCP version on the Internet, employs passive congestion indication. It makes worse the network congestion. Recently, Brakmo and Peterson have proposed a new version of TCP, which is named TCP Vegas, with a fundamentally different congestion control scheme from that of the Reno. Many studies indicate that the Vegas is able to achieve better throughput and higher stability than the Reno. But there are two unfairness problems in Vegas. These problems hinder the spread of the Vegas in current Internet. In this paper, in order to solve these unfairness problems, we propose a new congestion control algorithm called TCP PowerVegas. The existing Vegas depends mainly only on the rtt(round trip time), but the proposed PowerVegas use the new congestion control scheme combined the Information on the rtt with the information on the packet loss. Therefore the PowerVegas performs the congestion control more competitively than the Vegas. Thus, the PowerVegas is able to solve effectively these unfairness problems which the Vegas has experienced. To evaluate the proposed approach, we compare the performance among PowerVegas, Reno and Vegas under same network environment. Using simulation, the PowerVegas is able to achieve better throughput and higher stability than the Reno and is shown to achieve much better fairness than the existing Vegas.

• Journal of KIISE:Information Networking
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• v.32 no.6
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• pp.732-744
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• 2005

Previous works for the AE(Assured Forwarding) service in the Diffserv network have no sufficient consideration on the fairness of bandwidth share based on the target rate and the effect or RTT and UDP. Also Previous works act like Best-effort service in the UPN(under-Provisioned Network) condition. In this paper, in order to solve these problems, we propose the PFDSA(Proportionally Fair Differentiated Service Architecture) composed of tmTRA3CM(tcp-microflow based Target rate and an Aware Three color Marking), um3CM(udp-microflow based Three color Marker), TRBD(Target Rate Based Dropper), and target rate adjusting function. In the results of comparing the performance among existing mechanisms and the PFDSA, the PFDSA was able to mitigate the RTT and UDP effect better than the former. The PFDSA was shown to provide good performance for transmission rates proportional to various target rates in the UPN condition.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1B
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• pp.39-46
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• 2005

As the multimedia application traffic takes more portion in the internet traffic, it is necessary to control the network congestion through the congestion control protocol. In addition, the QoS-enabled networks such as DiffServ become an indispensable technology when running the multimedia applications. However, the previously proposed end-to-end congestion control algorithms take the round trip time to react the network congestion. Thus, as the RTT becomes larger, the reaction against the congestion gets delayed further, while the network congestion gets worse. In addition the performance of end-to-end congestion control algorithm is degraded if the QoS-enabled network runs the congestion control mechanism in the network level without any coordination between them. In this paper, we propose the early rate adaptation protocol for the DiffServ network which effectively linke the congestion control algorithm at the host and the congestion mechanism in the network together. By taking advantage of early congestion notification from the network it is possible to react the network congestion more quickly and effectively.

• The Transactions of the Korea Information Processing Society
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• v.6 no.1
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• pp.148-158
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• 1999

ABR traffic form ATM LAN is controlled through RM cell, and the interface function to public ATM network is necessary to provide ABR service efficiently. This paper presnets new interface architecture, which is based on "Projected Node" [6]. using AIPU(ABR Interface Proxy Unit) to support ABR traffic streams incoming from ATM LAN in the public ATM network. For the efficient interworking, the AIPU has designed for interworking functions with ERICA switch mechanism. Conventional ERICA switch mechanism specified in TM 4.0 is basically used for short distance comparative to public network, however AIPU adopts the novel control mechanism to cover logng roud trip time (RTT). To improve the problems and to provide a dynamic range of UCI(Update Count Interval), this paper proposes, a novel control scheme, DUCI ( Dynamic Update Count Interval. And the paper shows inefficiencies of ERICA mechanism with fixed UCI through the simulation results, and represents the performance enhancement of DUCI mechanism developed to adjust the update count interval dynamically.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.1
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• pp.79-95
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• 2016

In order to improve the Quality-of-Service (QoS) of latency sensitive applications in next-generation cellular networks, multi-path is adopted to transmit packet stream in real-time to achieve high-quality video transmission in heterogeneous wireless networks. However, multi-path also introduces two important challenges: out-of-order issue and reordering delay. In this paper, we propose a new architecture based on Software Defined Network (SDN) for flow aggregation and flow splitting, and then design a Multiple Access Radio Scheduling (MARS) scheme based on relative Round-Trip Time (RTT) measurement. The QoS metrics including end-to-end delay, throughput and the packet out-of-order problem at the receiver have been investigated using the extensive simulation experiments. The performance results show that this SDN architecture coupled with the proposed MARS scheme can reduce the end-to-end delay and the reordering delay time caused by packet out-of-order as well as achieve a better throughput than the existing SMOS and Round-Robin algorithms.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.1
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• pp.50-55
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• 2021

Multi-path system is a system in which utilizes various networks simultaneously. It is expected that multi-path system can enhance communication speed, reliability, security of network. In this paper, we focus on path selection in multi-path system. To select optimal path, we propose deep reinforcement learning algorithm which is rewarded by the round-trip-time (RTT) of each networks. Unlike multi-armed bandit model, deep Q learning is applied to consider rapidly changing situations. Due to the delay of RTT data, we also suggest compensation algorithm of the delayed reward. Moreover, we implement testbed learning server to evaluate the performance of proposed algorithm. The learning server contains distributed database and tensorflow module to efficiently operate deep learning algorithm. By means of simulation, we showed that the proposed algorithm has better performance than lowest RTT about 20%.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.99-101
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• 2011

본 논문에서는 WiMAX 의 신호세기(CINR: Carrier to Interference + Noise Ratio)의 변화와 예측 방법에 대해 연구한다. WiMAX 와 같은 무선 네트워크에서는 신호세기가 불규칙하게 변화하기 때문에 비디오 스트리밍시 에러나 지연이 발생할 수 있다. CINR 은 클라이언트 측의 가용 대역폭을 결정하는 중요한 요소이다. 하지만 기존의 RTCP(Real-time Transport Control Protocol)를 이용한 비트율 제어는 RTT(Round Trip time), PLR(Packet Loss Rate) 등의 정보를 사용하기 때문에 부정확하거나 지연이 발생될 수 있다. 이를 보완하기 위한 CINR 을 직접 비트율 제어에 사용하는 방법에 대해 연구하기에 앞서, 본 논문에서는 CINR 을 분석하고 미래의 값을 예측하는 방법에 대해 연구한다. 본 논문의 분석을 통해 CINR 의 예측을 비교적 정확하게 수행할 수 있다면 앞으로의 가용 대역폭을 비교적 정확하게 예측할 수 있고 효율적인 비디오 스트리밍 시스템을 제안할 수 있다.