• Proceedings of the Korean Information Science Society Conference
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• 2005.07a
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• pp.361-363
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• 2005

최근 컴퓨터 기술의 발전과 더불어 급속하게 발전하는 네트워크 기술이 실생활에 보급되고 네트워크를 통한 멀티미디어 데이터의 교환이나 전송과 같은 서비스들이 활성화 되면서 멀티미디어 데이터가 점점 대용량화 다양화되고 있다. 기존의 멀티미디어 서비스는 정해진 서버로부터 미리 데이터를 받아서 보는 다운로드 서비스가 대부분이었고 스트리밍 서비스라 할지라도 사용자의 기호나 원하는 요구사항에는 미치지 못 하였다. 지금까지 스트리밍 데이터를 전송 하는 프로토콜로 주로 UDP(User Datagram Protocol)를 사용하였다. 하지만 UDP는 혼잡제어를 하지 않으며 현재 인터넷의 주요 트래픽인 TCP(Transmission Control Protocol) 트래픽은 혼잡제어를 한다. 그래서 UDP에 의한 스트리밍 서비스는 TCP 트래픽의 전송률을 저하시키며 더 나아가 네트워크의 전체의 성능을 저하시키는 요인이 될 수 있다. 본 논문에서는 IETF(Internet Engineering Task Force)에서 제정한 실시간 스트리밍 데이터 서비스를 위한 표준인 RTP(Real-Time Transport Protocol)/RTCP(Real-Time Transport Control Protocol)를 적용하여 RTCP의 정보를 가지고 현재 네트워크 상태를 판단하고 스트리밍 서비스를 할 때 데이터의 전송률은 TCP 친화적인 전송률로 조절하는 스트리밍 서비스를 구현하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.3B
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• pp.487-492
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• 2000

The ABR(available bit rate) service has been developed to support data application over ATM(asynchronous Transfer mode) networks, and TCP is the most widely used transport layer protocol in existing data networks. Thus, the performance improvement of TCP over ABR is an issue of cardinal importance. In this paper, we propose algorithm of the adjustment of congestion window size utilizing RM(resource Management) cell information. The motivation is to use more reliable information in ER(explicit rate) field within RM cell to determine TCP window size. Simulations are performance using the suggested algorithm is improved, as compared to TCP-reno.

• Journal of KIISE:Information Networking
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• v.33 no.5
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• pp.369-379
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• 2006

In recent years, there have been many researches about Ad-hoc Networks which is available to communicate freely between mobile devices by using multi-hop without any support of relay base or access point. TCP that used the most widely transport protocol in the Internet repeats packet loss and retransmission because it increases congestion window size by using reactive congestion control until packet loss occurs. As a result of this, energy of mobile device is wasted unnecessarily. In this paper, we propose TCP-New Veno in order to improve the energy efficiency of mobile device. According to the state of network, the scheme adjusts appropriate size of congestion window. Therefore, the energy efficiency of mobile device and utilization of bandwidth are improved by the scheme. From the simulation by using ns-2, we could see more improved energy efficiency with TCP-New Veno than those with TCP in Ad-hoc Networks.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.650-659
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• 2004

Domestic wireless network environment is changing rapidly while adapting to meet service requirements of users and growth of market. As a result, reliable data transmission using TCP is also expected to increase. Since TCP assumes that it is used in wired networt TCP suffers significant performance degradation over wireless network where packet losses are not always result of network congestion. Especially RTO imposes a great performance degradation of TCP. In this paper, we propose DAC$^{＋}$ and EFR in order to prevent performance degradation by quickly detecting and recovering loss without RTO during fast recovery. Compared with TCP NewReno, proposed scheme shows improvements in steady-state in terms of higher fast recovery Probability and reduced response time.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.1
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• pp.414-435
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• 2017

As network adaptive streaming technology becomes increasingly common, transport protocol also becomes important in guaranteeing the quality of multimedia streaming. At the same time, because of the appearance of high-quality video such as Ultra High Definition (UHD), preventing buffering as well as preserving high quality while deploying a streaming service becomes important. The Internet Engineering Task Force recently published Multipath TCP (MPTCP). MPTCP improves the maximum transmission rate by simultaneously transmitting data over different paths with multiple TCP subflows. However, MPTCP cannot preserve high quality, because the MPTCP subflows slowly increase the transmission rate, and upon detecting a packet loss, drastically halve the transmission rate. In this paper, we propose a new multipath congestion control scheme for high-quality multimedia streaming. The proposed scheme preserves high quality of video by adaptively adjusting the increasing parameter of subflows according to the network status. The proposed scheme also increases network efficiency by providing load balancing and stability, and by supporting fairness with single-flow congestion control schemes.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.11
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• pp.4068-4086
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• 2014

Due to the prevalence of powerful mobile terminals and the rapid advancements in wireless communication technologies, the wireless video streaming service has become increasingly more popular. Recent studies show that video streaming services via Transmission Control Protocol (TCP) are becoming more practical. TCP has more advantages than User Diagram Protocol (UDP), including firewall traversal, bandwidth fairness, and reliability. However, each video service shares an equal portion of the limited bandwidth because of the fair sharing characteristics inherent in TCP and this bandwidth fair sharing cannot always guarantee the video quality for each user. To solve this challenging problem, an Adaptive Multiple TCP (AM-TCP) scheme is proposed in this paper to guarantee the video quality for mobile devices in wireless networks. AM-TCP adaptively controls the number of TCP connections according to the video Rate Distortion (RD) characteristics of each stream and network status. The proposed scheme can minimize the total distortion of all participating video streams and maximize the service quality by guaranteeing the quality of each video streaming session. The simulation results show that the proposed scheme can significantly improve the quality of video streaming in wireless networks.

• The KIPS Transactions:PartC
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• v.11C no.2
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• pp.245-252
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• 2004

Stream Control Transmission Protocol(SCTP) is a new connection-oriented, reliable delivery transport protocol operating on top of an unreliable connectionless packet service such as IP. It inherits many of the functions developed for TCP, including flow control and packet loss recovery functions. In addition, it also supports transport layer multihoming and multistreaming In this paper, we study the impact of multi-homing on the performance of SCTP. We first compare performance of single-homed SCTP. multi-homed SCTP, TCP Reno and TCP SACK. We, then describe potential flaw in the current SCTP retransmission policy, when SCTP host is multihomed. Our Results show that SCTP performs better than TCP Reno and TCP SACK due to several changes from TCP in its congestion control mechanism. In particular. multi-homed SCTP shows the best result among the compared schemes. Through experimentation for multi-homed SCTP, we found that the current SCTP retransmission policy nay deteriorate the perfomance when the retransmission path it worse than the original path. Therefore, the condition of retransmission path is a very important factor In SCTP performance and a proper mechanism would be required to measure the condition of the retransmission path.

• Proceedings of the IEEK Conference
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• 2000.06a
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• pp.275-278
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• 2000

Recently ATM(Asynchronous Transfer Mode) technology is facing challenges from Integrated Service IP(Internet Protocol), IP router, Gigabit Ethernet. Although ATM is approved by ITU-T as the standard technology in B-ISDN, its survival is still in question. In the ATM networks, the Guaranteed Frame Rate(GFR) service has been designed to accommodate non-real-time applications, such as TCP(Transmission Control Protocol)/IP based traffic. The GFR service not only guarantees a minimum throughput at the frame level, but also supports a fairshare of available resources. We have studied different discarding and scheduling schemes, and compared their throughput and fairness when TCP/IP Traffic is carried. Through simulations, we know that only per-VC queueing with weighted Round Robin(WRR) can guarantee Minimum Cell Rate Among all the Schemes that have been experimented, we recommend DT-EPD(Dynamic Threshold-Early Packet Discard) integrated with MCRplus(Minimum Cell Rate) to support the GFR service.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.3B
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• pp.130-137
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• 2005

Wireless communication environment is changing rapidly as we use new wireless communication technology such as WiBro to access high speed Internet. As a result, reliable data transmission using TCP is also expected to increase. Since TCP assumes that it is used in wired network, TCP suffers significant performance degradation over wireless network where packet losses are related to non-congestion loss. Especially RTO imposes a great performance degradation of TCP. In this paper, we analyze the loss recovery probabilities based on previous researches, and use simulation results of our algorithm to show that it prevents performance degradation by quickly detecting and recovery losses without RTO during fast recovery.

• 제어로봇시스템학회:학술대회논문집
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• 2000.10a
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• pp.118-118
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• 2000

This paper suggests the methodology to guarantee the rea1-time service over Ethernet and TCP/IP and to solve the problems such as re-transmission due to collision and platform-transparency and independence. This paper proposes a new mechanism called Real-time Ethernet protocol combined the centralized medium control mechanism with the token passing mechanism. The centralized medium control mechanism is used to get the token, the right to transmit the data, and the token passing mechanism is used to return the token to the token controller and to transmit data within the specified time interval. The proposed Real-Time Ethernet Protocol is based on Java and Java RMI(Remote Method Invocation). The presented protocol is believed to work enough in real-time applications considering latency occurred due to the Java RMI, which is less small than data transmission time.