• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.10a
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• pp.367-368
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• 2015

This paper evaluates performance degradation of an NG-EPON (1next-generation Ethernet-passive opticla network) system due to an unused remainder of a grant. Since a packet segmentation is not permitted in NG-EPON, a grant is wasted if the grant size is less than the packet size. Using simulations, we evaluate performance degradation due to the unused remainder.

• The KIPS Transactions:PartC
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• v.8C no.3
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• pp.335-344
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• 2001

최근 인터넷 환경이 반도체, 광통신 그리고 차세대 인터넷 기술의 발달로 고성능화 되어가고 있다. 따라서 고성능 인터넷을 위한 TCP의 성능 향상 연구가 매우 중요해졌다. 그러나 기존 TCP는 수신위도 버퍼의 물리적 크기에 의하여 최대 전송 성능과 대역폭 탐색 기능이 제한을 받는 구조적인 문제점을 갖고 있다. 본 논문에서는 이를 해결하기 위하여 수신 호스트에 가상 윈도 개념을 도입하였다. 이는 송신 호스트가 RTT 동안 균일하게 세그먼트를 분산시켜서 패킷을 전송할 때 세그먼트 간격 시간 동안 수신 호스트의 처리 능력을 가상윈도로 나타내는 것이다. 따라서 가상 윈도의 크기는 수신 호스트의 성능에 비례하기 때문에 수신 호스트가 고성능일 경우 TCP의 전송 능력 성능이 더 높아질 수 있다. 초고속 인터넷일 경우 제안 알고리즘이 기존 TCP보다 전송능력에 있어 1.5∼5배 개선되는 것을 네트워크 시뮬레이션인 NS2를 이용하여 확인하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2003.04d
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• pp.512-514
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• 2003

본 논문은 모바일 환경에서 제한된 메모리를 가지고 있는 수신자에 의 한 TCP흐름 제어 방법을 제안한다. TCP 흐름 제어는 송신자에서 수신자에게 전달되는 Advertised 윈도우 크기를 조정하여 수행된다. 수신자는 무선 대역폭과 종단간 패킷 왕복 시간을 동적으로 측정하며 최적의 Advertised 윈도우 크기를 계산하고 송신자의 전송률을 무선 대역폭으로 제한한다. 제안된 흐름 제어 기법은 제한된 메모리를 가진 수신자를 고려하였으며 무선 네트웍의 특성을 고려 한 효율적 인 TCP 흐름 제어로 TCP의 전송 성능 향상과 종단간 패킷 왕복 시간의 지연을 줄일 수 있도록 하였다. 제안된 흐름 제어 기 법의 효율성과 성능을 구현과 실험을 통해 검증한다.

• The KIPS Transactions:PartC
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• v.12C no.1 s.97
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• pp.121-128
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• 2005

Tho existing TCP(Transmission Control Protocol) is known to be unsuitable for a network with the characteristics of high RDP(Bandwidth-Delay Product) because of the fixed small or large buffer size at the TCP sender and receiver. Thus, some trial cases of adjusting the buffer sizes automatically with respect to network condition have been proposed to improve the end-to-end TCP throughput. ATBT(Automatic TCP fluffer Tuning) attempts to assure the buffer size of TCP sender according to its current congestion window size but the ATBT assumes that the buffer size of TCP receiver is maximum value that operating system defines. In DRS(Dynamic Right Sizing), by estimating the TCP arrival data of two times the amount TCP data received previously, the TCP receiver simply reserves the buffer size for the next arrival, accordingly. However, we do not need to reserve exactly two times of buffer size because of the possibility of TCP segment loss. We propose an efficient TCP buffer tuning technique(called TBT-PLR: TCP buffer tuning algorithm based on packet loss ratio) since we adopt the ATBT mechanism and the TBT-PLR mechanism for the TCP sender and the TCP receiver, respectively. For the purpose of testing the actual TCP performance, we implemented our TBT-PLR by modifying the linux kernel version 2.4.18 and evaluated the TCP performance by comparing TBT-PLR with the TCP schemes of the fixed buffer size. As a result, more balanced usage among TCP connections was obtained.

• Journal of the Korea Computer Industry Society
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• v.3 no.5
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• pp.631-640
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• 2002

The internet phones that are currently available in use adopt packet exchange system, transferring through various routes and lacking sufficient band width with a result that there is an accompanied delay for packet transmission since the traffic is increased, accordingly affecting a lot in sound quality and speed. Two solutions for such troubles are suggested in this study to improve sound quality of internet phones. Firstly, we minimize the delay and damage regarding packet size based on traffic size by using the data algorithm from variable packets in order to supplement decreased sound quality due to the delay and damage of sound data. The second suggestion is to employ a method of Jitter compensation by giving an appropriate initial delay time with regenerating buffers to bypass troubles from Jitter, From employing the Jitter compensation method, we found that there is a sound quality improvement due to the less stoppage phenomenon.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.12A
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• pp.1320-1328
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• 2007

In recent year, the popularity of WLAN has generated much interests on improvement and performance analysis of the IEEE 802.11 protocol. In this paper, we analyze two medium access methods of the IEEE 802.11 MAC protocol by investigating the MAC layer packet service times when arrival packet sizes have a general probability distribution. We use the M/G/1/K queueing model to analyze the throughput and the delay performance of IEEE 802.11 MAC protocol in a wireless LAN. We compare the performances of Basic access method and RTS/CTS access method. We take some numerical examples for the system throughput and the queue dynamics including the mean packet delay and packet blocking probability.

• The Journal of the Korea Contents Association
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• v.3 no.1
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• pp.21-30
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• 2003

Retracing schemes using packet marking are currently being studied to protect network resources by isolating DDoS attack. One promising solution is the probabilistic packet marking (PPM). However, PPM can't use ICMP by encoding a mark into the IP identification field. Likewise, it can't identify the original source through a hash function used to encode trace information and reduce the mark size. In addition, the retracing problem overlaps with the result from the XOR operation. An algorithm is therefore proposed to pursue the attacker's source efficiently. The source is marked in a packet using a router ID, with marking information abstracted.

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.937-942
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• 2003

In this paper, we measured and examined RTT delays and packet losses according to the changes of stationary loads for two typical stream-type traffics, a DV and a MPGE2 on the R＆D Gigabit Network testbed, JGN. As the result of our actual measurements, we realized that the packet size of stationary load have no effects on a DV and a MPGE2 stream on the very high-speed network(50Mbps, IP over ATM). When its bandwidth and stationary load exceeds 95％ of network bandwidth, packet losses appeared and RTT delay increased rapidly. Also we realized that the number and size of Receive ＆ Transmit buffer on the end systems have no effects on packet losses and RTT delays.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.218-223
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• 2008

In this paper, we implement several methods to improve an audio collaboration system over IP networks, and then evaluate the performance of the implemented methods. In general, speech and audio quality degrades depending on the characteristics of IP networks such as jitter and packet loss. In order to reduce this quality degradation, we propose a lower bit rate audio delivery scheme using the MPEG-2 AAC (Advanced Audio Coding) audio codec in a viewpoint that a packet loss rate could be reduced by a smaller packet size. In addition, iLBC (Internet Low-Bitrate Codec) and the G.711 packet loss concealment algorithm defined by IEFT and ITU-T, respectively, are applied to a audio collaboration system. RAT (Robust-Audio Tool)[7] is used as a baseline platform for the implementation of the proposed methods. It is shown from the implementation that the implemented MPEG-2 AAC audio codec with a bitrate of 256 kbit/s performs as similar as the uncompressed audio quality with a bitrate of 512 kbit/s, and that iLBC and the G.711 packet loss concealment algorithm can improve speech quality when a packet loss rate is 2~10%.

• Journal of Internet Computing and Services
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• v.3 no.6
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• pp.43-51
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• 2002

In this paper, we study the new packet dropping scheme using an active queue management algorithm. Active queue management mechanisms differ from the traditional drop tail mechanism in that in a drop tail queue packets are dropped when the buffer overflows, while in active queue management mechanisms, packets may be dropped early before congestion occurs, However, it still incurs high packet loss ratio when the buffer size is not large enough, By detecting congestion and notifying only a randomly selected fraction of connection, RED causes to the global synchronization and fairness problem. And also, it is the biggest problem that the network traffic characteristics need to be known in order to find the optimum average queue length, We propose a new efficient packet dropping method based on the active queue management for congestion control. The proposed scheme uses the per-flow rate and fair share rate estimates. To this end, we present the estimation algorithm to compute the flow arrival rate and the link fair rate, We shows the proposed method improves the network performance because the traffic generated can not cause rapid fluctuations in queue lengths which result in packet loss