• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.43 no.10 s.352
• /
• pp.8-17
• /
• 2006

Recently, the high-speed Internet users increase rapidly and broadband networks have been widely deployed. However, the current TCP congestion control algorithm was designed for relatively narrowband network environments, and thus its performance is inefficient for traffic transport in broadband networks. To remedy this problem, the TCP having an enhanced congestion control algorithm is required for broadband networks. In this paper, we propose an improved TCP congestion control that can sufficiently utilize the large available bandwidth in broadband networks. The proposed algorithm predicts the available bandwidth by using ACK information and RTT variation, and prevents large packet losses by adjusting congestion window size appropriately. Also, it can rapidly utilize the large available bandwidth by enhancing the legacy TCP algorithm in congestion avoidance phase. In order to evaluate the performance of the proposed algorithm, we use the ns-2 simulator. The simulation results show that the proposed algorithm improves not only the utilization of the available bandwidth but also RTT fairness and the fairness between contending TCP flows better than the HSTCP in high bandwidth delay product network environment.

• The Journal of the Korea Contents Association
• /
• v.5 no.5
• /
• pp.248-254
• /
• 2005

The existing buffer management scheme is caused by with burstness characteristic of the TCP traffic and with only the transmission which is not loss it provides the smallest transmission rate guarantee where the GFR demands and a fair characteristic. In order to provide a high fair characteristic from the dissertation which it sees with the smallest transmission rate guarantee where the GFR demands it proposed the existing buffer algorithm which applies a Fuzzy mechanism in the existing buffer management technique. The proposed algorithm decides a packet disuse used by three parameters which are composed of tagging information, the buffer usage, and the load of VC. Simulation results shows that the fairness and goodput of the proposed algorithm were excellent where the size of MCR will become larger, from the switch the Double-EPD or the DFBA was visible a similar efficiency even from size change of the buffer. The algorithm which is proposed provides a good throughput and a fair characteristic.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2018.11a
• /
• pp.18-21
• /
• 2018

최근 시청자에게 실재적인 몰입감을 제공하는 실감미디어가 발전하고 있다. 이러한 실감미디어 중 가장 접근성이 뛰어난 것은 VR으로, 현재 다양한 서비스에 응용되고 있는 상황이다. 하지만 360VR은 높은 비트 전송률과 고해상도의 특성을 지니고 있기 때문에 대역폭 비용 및 시청자의 QoE 보장의 불확실성 등의 다양한 문제가 있다. 이러한 문제를 개선시키기 위해 HTTP 기반의 적응적 스트리밍 기법이 발전해나가고 있으며 이 기술의 표준으로 MPEG-DASH가 채택되었다. MPEG-DASH는 TCP 기반의 전송 프로토콜을 사용하고 있지만 현재 TCP는 고용량의 데이터가 전송되는 웹 기반에서 HTTP의 병목현상을 일으켜 대역폭 효율성을 떨어뜨리는 하나의 원인으로 나타나고 있다. 이러한 문제를 해결하기 위해 본 논문에서는 UDP 기반의 QUIC 프로토콜을 MPEG-DASH에 적용하는 시스템을 고안하고 이를 QUIC-DASH라 칭한다. 고해상도의 360VR 송출 실험으로 QUIC-DASH 시스템과 기존의 MPEG-DASH 시스템을 비교 분석한 결과로 대역폭의 절감이 이루어진 것을 확인하였다.

• Journal of KIISE:Information Networking
• /
• v.29 no.3
• /
• pp.233-241
• /
• 2002

In this paper, we proposed a new mechanism that solves the fairness problem between unicast traffic using the TCP and multicast traffic using the UDP, and satisfies the requirement of various receivers fairly in the Internet. The proposed mechanism decentralizes the load of blanket transmission rate control from sender to designated server, and uses the method that talc designated server intercepts the sender's data and controls the transmission rate suitable for it's local network. Therefore, the proposed mechanism not only provides multicast service by accurate estimation of the network status of each receiver, but also realizes both the inter-session fairness and the intra-session fairness problem.

• 대한공업교육학회지
• /
• v.33 no.2
• /
• pp.287-299
• /
• 2008

HTTP is one of the most widely used protocols of the WWW. Currently it uses TCP as the transport layer protocol to provide reliability. The HTTP uses separate TCP connection for each file request and adds unnecessary head-of-line blocking overhead for the file retrieval. The web application is short sized and affected by the increased handover latency of TCP in wireless environment. SCTP has attractive features such as multi-streaming and multi-homing. SCTP's multi-streaming and multi-homing avoid head-of-line blocking problem of TCP and reduce handover latency of TCP in wired and wireless environment. Mean response time is the important measure in most web application. In this paper, we present the comparison of mean response time between HTTP over SCTP with that of HTTP over TCP in wired and wireless environments using NS-2 simulator. We measured mean response time for varying packet loss rate, bandwidth, RTT, and the number of web objects in wired environment and mean response time and packet loss rate for varying moving speed and region size in wireless environment. Our experimental result shows that SCTP reduces the mean response time of TCP based web traffic.

• Journal of KIISE:Information Networking
• /
• v.37 no.5
• /
• pp.363-373
• /
• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

• Proceedings of the IEEK Conference
• /
• 2000.11a
• /
• pp.17-20
• /
• 2000

TCP is ill-suited to real-time multimedia applications. Its bursty transmission, and abrupt and frequent wide rate fluctuations cause high delay jitters and sudden quality degradation of multimedia applications. Deploying non congestion controlled traffic results in extreme unfairness towards competing TCP traffic. Therefore, they need to be enhanced with congestion control schemes that not only am at reducing loss ratios and improve bandwidth utilization but also are fair towards competing TCP connections. This paper proposes a differentiated rate adaptation algorithm based on loss and round trip time. Rate in a sender quickly responds to loss ratio and holds steady state. Additionally, this algorithm reduces loss ratio by loss prediction in a receiver.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.5 no.6
• /
• pp.1078-1082
• /
• 2001

This is an example of ABSTRACT format. We made a HTTP server using 8 bit microprocessor It was TMP84c015 which applied a z80 core and RTL8019AS was installed for an ethernet physical layer. Assembly language was used to optimized a performance of the MPU, to overcome an restriction of memory sire and to maximize the throughput of packet using TCP, UDP, IP, ICMP, ARP protocol. We used LabVIEW to verify the each protocol on the client side.

• Proceedings of the Korea Information Processing Society Conference
• /
• 2005.05a
• /
• pp.1569-1572
• /
• 2005

인트라넷은 TCP 를 지원하는 기존의 LAN 환경에서 인터넷 기술을 그대로 채택하였다. 이는 인터넷 기반의 TCP Parameter 값을 그대로 인트라넷에 적용하였음을 의미하며 서로 다른 환경에서 같은 Parameter 값의 적용은 네트워크 성능 저하의 원인이 되기도 한다. 네트워크의 성능을 확인하는 방법으로 Rtt(Round Trip Time:응답시간)와 Thput(Throughput:처리률)이 있다. 이값은 전송 거리와 트래픽 양에 따라 다르게 나타난다. 따라서 본 논문에서는 LAN 기반으로 전송거리에 따른 즉, Topology diameter 와 TCP Parameter 와의 관계를 RTT 와 Thput 을 통해 분석해 보았다.

• Journal of KIISE:Information Networking
• /
• v.30 no.2
• /
• pp.233-243
• /
• 2003

With the original Transmission Control Protocol(TCP) design, which is particularly targeted at the wired networks, a packet loss is assumed to be caused by the network congestion. In the wireless environment where the chances to lose packets due to transmission bit errors are not negligible, though, this assumption may result in unnecessary TCP performance degradation. In this paper, we propose three schemes that improve the ability to conceal the packet losses in the wireless network while limiting the degree of violating TCP end-to-end semantics to a temporary incidents. If there happens a packet loss at the wireless link and there is a chance that the loss is noticed by the sending TCP, the proposed schemes send an indirect acknowledgement. Each of the proposed schemes uses different criteria to decide whether there is a chance that the packet loss occurred in the wireless part is noticed by the sender. In order to limit the buffer overhead in the base, the indirect acknowledgements are issued only when the length of buffer is less than a certain threshold. We use simulation to compare the overhead and the performance of the proposed schemes, and to show that the proposed schemes improve the TCP performance compared to Snoop with a limited amount of buffer at the base station.