• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.4B
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• pp.227-233
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• 2007

According to the recent development of network technology, the demands of users are diversified and the needs of multimedia traffic are increasing. In general, UDP(User Datagram Protocol) traffic is used to transport multimedia data, which satisfied the real-time and isochronous characteristics. UDP traffic competes with TCP traffic and incur the network congestion. However, TCP traffic performs network congestion control but does not consider the receiver's status. Thus, it is not appropriate in case of streaming services. In this paper, we solve a fairness problems and proposed a network algorithm based on RTP/RTCP(Real-time Transport Protocol/Realtime Transport Control Protocol) in view of receiver status. The POBA(Priority Ordered Buffer Algorithm), which applies priorities in the receiver's buffer and networks, shows that it provides the appropriate environment for streaming services in view of packet loss ratio and buffer utilization of receiver's buffer compared with the previous method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.12B
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• pp.800-808
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• 2007

With the development of wireless data communication technology, all IP-based network will become compositions of wired and wireless networks. TCP is a connection-oriented, reliable transport protocol and has been used as de facto standard in most wired networks. Because TCP's congestion control algorithm could not distinguish congestion from BER, link failure and frequent route changes, TCP shows a poor performance over mobile ad-hoc networks. In this paper, the theoretical feature of TCP was studied and the performance of TCP over mobile ad-hoc networks was analyzed with ns2.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.7A
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• pp.700-709
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• 2007

The heterogeneous wireless access networks has been envisioned to characterize the future wireless networks. In such environments, TCP(Transmission Control Protocol) has to experience poor end-to-end performance because bandwidth and link delay change suddenly when a mobile node moves over different types of wireless networks, which is called vertical handoff. In this paper, we propose a new TCP which maintains each set of congestion control variables, which we call TCP context, for each type of wireless network. The proposed TCP can switch the TCP context against vertical handoff in order to adjust quickly to a newly arrived network. In simulations, the proposed TCP has higher throughput than TCP SACK(Selective Acknowledgment Options) due to its great features to vertical handoff situations.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.297-312
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• 2005

The multimedia traffic of continuous video and audio data via streaming service accounts for a significant and expanding portion of the Internet traffic. This streaming data delivery is mostly based on RTP with UDP. However, UDP does not support congestion control. For this reason, UDP causes the starvation of congestion controlled TCP traffic which reduces its bandwidth share during overload situation. In this paper, we propose a new TCP-friendly rate control scheme called 'TF-RTP(TCP-Friendly RTP)'. In the congested network state, the TF-RTP exactly estimates the competing TCP's throughput by using the modified parameters. Then, it controls the sending rate of the video streams. Therefore, the TF-RTP adjusts its sending rate to TCP-friendly and fair share with competing TCP traffics. Through the simulation, we prove that the TF-RTP correctly estimates the TCP's throughput and improves the TCP-friendliness and fairness.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.660-666
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• 2004

Overall TCP performance represented by end-to-end throughput is largely dependent upon its loss recovery performance. In particular non-congestion packet losses caused by transmission errors degrade TCP performance seriously. Using Markov process, we analyze TCP loss recovery performance for correlated packet losses caused by multipath fading. The results show that loss recovery performance can be severely affected by burstiness in packet losses, even if overall packet loss ratio is very low.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.05b
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• pp.104-108
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• 2003

모바일에서의 TCP 데이터전송은 무선 고유의 특성인 높은 에러율로 인하여 TCP의 성능에 많은 장애를 가져 오고있다. 특히, 셀간 핸드오프 등으로 인해 TCP의 빈번한 혼잡제어가 발생하므로서 대역폭 저하가 발생할 수 있다. 본 논문에서는 대역폭 저하가 발생하는 것을 억제하기 위하여 모바일 노드가 높은 에러 발생지역에 위치한 경우 시그널링을 이용하여 패킷 전송을 중지시키는 방법으로 핸드오프 영역에서 효율적인 전송이 이루어지도록 함으로써 TCP의 성능을 향상시킬 수 있는 방법을 제안하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2008.06d
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• pp.137-140
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• 2008

TCP는 유선망에 최적화되도록 변화 된 결과, 무선망에서는 무선망의 단점인 비연속적인 연결 링크로 인해 전송손실이 발생하게 된다. 그러나 어플리케이션 프로그램에서는 이러한 문제로 인해 발생되는 오류를 네트워크의 혼잡으로 인한 손실로 오인하게 된다. 그 결과 어플리케이션 프로그램에서는 혼잡제어 메커니즘이 수행되어 전송율을 줄이므로 네트워크 성능이 저하되는 문제점을 초래한다. 이러한 이유로 최근 유무선 혼합망에서 TCP의 성능을 향상시키기 위한 연구가 활발히 이루어지고 있다. 본 논문에서는 기존에 제안된 개선 모델들 중 상대적으로 많이 사용되고 있는 TCP Westwood에 대해서 성능을 보완하는 기법을 제안하고자 한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2013.11a
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• pp.323-326
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• 2013

최근 지능형 교통 시스템 (ITS) 에 대한 관심이 증가함에 따라 ITS 응용을 지원해 줄 수 있는 차량 에드혹 네트워크 (VANET, Vehicular Ad-Hoc Network) 기술이 주요 이슈로 대두되고 있다. 특히 트랜스포트 계층에서는 종단간 신뢰성이 요구되는 ITS 응용들을 지원하기 위해 TCP를 적용할 필요가 있다. 하지만 VANET 환경에서 TCP는 혼잡 제어 기능의 오동작, 이동성에 의한 경로 재설정 오버헤드, 경쟁(contention) 등으로 인해 throughput을 불필요하게 감소시킬 수 있다. 따라서 VANET 환경에서 TCP 성능을 향상시킬 수 있는 다양한 기법들에 대한 연구가 진행되고 있다. 본 논문에서는 이러한 기법들을 크게 혼잡 탐색 기법, cross-layer 기법으로 나누어 대표적인 두 가지 기법들을 각각 비교, 분석한다.

• Journal of Internet Computing and Services
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• v.8 no.1
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• pp.15-23
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• 2007

Wireless LAN (IEEE 802.11) uses traditional TCP for reliable data transmission, But it brings the unintentional packet loss which is not congestion loss caused by handoff, interference, and fading in wireless LAN. In wireless LAN, TCP experiences performance degradation because it consumes that the cause of packet loss is congestion, and it decrease the sending rate by activating congestion control algorithm. This paper analyzes that correlation of throughput and buffer size for wireless buffer tuning. We find MBT (Maximum Buffer Threshold) which does not increase the throughput through the analysis, For calculation of MBT, we experiment the throughput by using high volume music data which is creased by real-time performance of piano. The experiment results is shown that buffer tuing based on MBT shows 20.3%, 21.4%, and 45.4% throughput improvement under 5ms RTT, 10ms RTT, and 20ms RTT, respectively, comparing with the throughput of operation system default buffer size, In addition, we describe that The setting of TCP buffer size by exceeding MBT does not have an effect on the performance of TCP.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.8 s.350
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• pp.84-90
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• 2006

With the widespread of the wireless Internet and wireless LAN, different wireless technologies such as 3G cellular networks and WLAN will cooperate to support more users and applications with higher data rate over wider areas. When a mobile node moves around in the hybrid networks, it needs to perform seamless vertical handoffs between different wireless networks to provide high performance data transmission. When an application with TCP connection in a mobile node performs a vertical handoff, TCP performance is degraded due to packet losses even though it maintains the previous TCP state information during handoff, because 3G and WLAN have different available bandwidth. In this paper, we propose a new congestion control algorithm for vertical handoff to improve the TCP performance by measuring the rough end-to-end available bandwidth and calculating the slow-start threshold. By ns-2 simulation, we show that the proposed algorithm enhances the TCP performance during vertical handoffs compared to the previous algorithms.