• Journal of Communications and Networks
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• v.6 no.3
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• pp.235-244
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• 2004

There have been a lot of approaches to evaluate and predict transmission control protocol (TCP) performance in a numerical way. Especially, under the recent advance in wireless transmission technology, the issue of TCP performance over wireless links has come to surface. It is because TCP responds to all packet losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to-end performance in wireless and lossy systems. By several previous works, although it has been already proved that overall TCP performance is largely dependent on its loss recovery performance, there have been few works to try to analyze TCP loss recovery performance with thoroughness. In this paper, therefore, we focus on analyzing TCP's loss recovery performance and have developed a simple model that facilitates to capture the TCP sender's behaviors during loss recovery period. Based on the developed model, we can derive the conditions that packet losses may be recovered without retransmission timeout (RTO). Especially, we have found that TCP Reno can retransmit three packet losses by fast retransmits in a specific situation. In addition, we have proved that successive three packet losses and more than four packet losses in a window always invoke RTO easily, which is not considered or approximated in the previous works. Through probabilistic works with the conditions derived, the loss recovery performance of TCP Reno can be quantified in terms of the number of packet losses in a window.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.7
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• pp.45-51
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• 2009

We have presented a routing mechanism that selects a route by considering channel statuses in order to fast transfer delay-sensitive data in WSNs (Wireless Sensor Networks). The existing methods for real-time data transfer select a path whose latency is the shortest or the number of hops is the smallest. An algorithm to select a real-time transfer path based on link error rates according to the characteristic of wireless medium was also suggested. However, the propagation delay and retransmission timeout affected by link error rates are shorter than channel assessment time and backoff time. Therefore, the mechanism proposed in this paper estimated the time spent in using a clear channel and sending out a packet, which is based on channel backoff rates. A source node comes to select a route with the shortest end-to-end delay as a fast transfer path for real-time traffic, and sends data along the path chosen. We found that this proposed mechanism improves the speed of event-to-sink data transfer by performing experiments under different link error and channel backoff rates.

• Proceedings of the KIEE Conference
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• 2006.10c
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• pp.473-475
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• 2006

FieldBus communication systems in industrial wired and wireless network may cause the degradation of the TCP performance due to the racket loss. TCP is particularly targeted at the wired networks, a packet loss is assumed to be caused by the network congestion. As a result, the performance of TCP decreases significantly when used over networks that exhibit a high bit error rate. In order to solve this problem, this paper designs and implements the WFSnoop mechanism which offers a fast local retransmission. The proposed mechanism does not require any changes in customer premises. Base on the simulation in the wired and wireless network environment, we analyzed the performance of the WFSnoop mechanism.

• Journal of Korea Multimedia Society
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• v.3 no.6
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• pp.643-651
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• 2000

ATM network technology is generally used for the integration of multimedia services in high-speed Internet. ABR (Available Bit Rate) and UBR (Unspecified Bit Rate) service classes have been developed specifically to support data application. In this paper, through the result of simulation, we analyzed the effect of TCP data transmission performance by using FRR (Fast Retransmission and Recovery) and Nagle's Algorithm on the UBR service, and by adjusting EFCI switch parameter on the ABR service. As a result of our study, performance improvement of TCP over ATM network is observed by adjusting TCP parameters and setting of effective switch parameter.

• KIPS Transactions on Computer and Communication Systems
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• v.10 no.2
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• pp.29-38
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• 2021

The Internet including mobile networks is developing to overcoming the limitation of physical distance and providing or acquiring information from remote locations. However, the systems that use video as primary information require higher bandwidth for recognizing the situation in remote places more accurately through high-quality video as well as lower latency for faster interaction between devices and users. The emergence of the 5th generation mobile network provides features such as high bandwidth and precise location recognition that were not experienced in previous-generation technologies. In addition, the Mobile Edge Computing that minimizes network latency in the mobile network requires a change in the traditional system architecture that was composed of the existing smart device and high availability server system. However, even with 5G and MEC, since there is a limit to overcome the mobile network state fluctuations only by enhancing the network infrastructure, this study proposes a high-definition video streaming system in ultra-low latency based on the SRT protocol that provides Forward Error Correction and Fast Retransmission. The proposed system shows how to deploy software components that are developed in consideration of the nature of 5G and MEC to achieve sub-1 second latency for 4K real-time video streaming. In the last of this paper, we analyze the most significant factor in the entire video transmission process to achieve the lowest possible latency.

• Journal of KIISE:Computing Practices and Letters
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• v.8 no.3
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• pp.336-343
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• 2002

Nowadays, after appearance of wireless network the existent internet environment is changing into the united wire/wireless network. But the present TCP regards all of the packet losses on transmission as the packet tosses due to the congestion. When it is applied on the wireless path, it deteriorates the end-to-end TCP throughput because it regards the packet loss by handoff or bit error as the packet loss by the congestion and it reduces the congestion window. In this paper, for solving these problems we propose the method that controls the performance of TCP on the wireless link by extending ECN which is used as a congestion control mechanism on the existent wire link. This is the method that distinguished the packet loss due to the congestion from due to bit error or handoff on the wireless network, so it calls the congestion control mechanism only when there occurs the congestion in the united wire/wireless network.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.8
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• pp.1-6
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• 2002

In this paper, we considered the CDMA/TDD system suitable for high-speed packet data transmission such as Internet and multimedia services, and a sequential decoding scheme which enables fast decoding and retransmission requirement. In addition, we Proposed an improved FANO algorithm, which adopts the competition path in order to reduce the number of revisit nodes. The conventional FANO algorithm suffered from the drawback of much more revisit nodes. Furthermore, we analyzed the performance of the CDMA/TDD system with the sequential decoding scheme we proposed over multipath channel.

• Journal of Internet Computing and Services
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• v.1 no.2
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• pp.1-10
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• 2000

ATM network technology is generally used for the solution of integrating multimedia service in high-speed Internet. In Internet protocol based on ATM services, If single cell is lost in ATM layer, the entire TCP packet will be lost. Therefore, TCP performance will be degraded. In order to reduce cell loss, when congestion occur, UBR+EPD mechanism is proposed to improve the throughput in TCP over UBR, and ER scheme is suggested in TCP over ABR. In this paper, we analyzed the performance improvement effect of UBR+EPD with FRR (Fast Retransmission and Recovery), the adjusting EPD threshold parameter (R), and variation of MTU (Maximum Transport Unit) size. As a result, through the analysis of performance, we know that the improved throughput and fairness are shown by the proposed scheme.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.9
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• pp.4572-4586
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• 2019

Mobile Edge Computing (MEC) and Information-Centric Networking (ICN) are essential network architectures for the future Internet. The advantages of MEC and ICN such as computation and storage capabilities at the edge of the network, in-network caching and named-data communication paradigm can greatly improve the quality of video streaming applications. However, the packet loss in wireless network environments still affects the video streaming performance and the existing loss recovery approaches in ICN does not exploit the capabilities of MEC. This paper proposes a Deep Learning based Loss Recovery Mechanism (DL-LRM) for video streaming over MEC based ICN. Different with existing approaches, the Forward Error Correction (FEC) packets are generated at the edge of the network, which dramatically reduces the workload of core network and backhaul. By monitoring network states, our proposed DL-LRM controls the FEC request rate by deep reinforcement learning algorithm. Considering the characteristics of video streaming and MEC, in this paper we develop content caching detection and fast retransmission algorithm to effectively utilize resources of MEC. Experimental results demonstrate that the DL-LRM is able to adaptively adjust and control the FEC request rate and achieve better video quality than the existing approaches.

• The KIPS Transactions:PartC
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• v.11C no.2
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• pp.229-234
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• 2004

In this study, the performance is compared between RFC compatible normal TCP and several speed constraints Ignored TCP. To do these, the main algorithms that constraints the transmit rate of TCP are modified. We have modified TCP protocol stack in a Linux kernel to compare the speeds between the standard TCP and our modified TCP. We find that if the destination is short distance away from the source and packet error is scarce then the speed differences between normal and modified TCP nay be negligible. However, if the destination is far away from the source and slow start algorithm is not adopted then the transfer time for small file is different greatly In addition, if packet error occurred frequently, our modified TCP is faster than the standard TCP regardless of distance.