• 대한공업교육학회지
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• v.33 no.2
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• pp.248-258
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• 2008

Current Internet uses HTTP (Hyper Text Transfer Protocol) as an application layer protocol and TCP (Transmission Control Protocol) as a transport layer protocol to provide web service. SCTP (Stream Control Transmission Protocol) is a recently proposed transport protocol with very similar congestion control mechanisms as TCP, except the initial congestion window during the slow start phase. In this paper, we present a mathematical model of object transfer latency during the slow start phase for HTTP over SCTP and compare with the latency of HTTP over TCP. Validation of the model using experimental result shows that the mean object transfer latency for HTTP over SCTP during the slow start phase is less than that for HTTP over TCP by 11%.

• Journal of Communications and Networks
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• v.12 no.3
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• pp.275-284
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• 2010

The current multihome-aware protocols (like stream control transmission protocol (SCTP) or parallel TCP for concurrent multipath data transfer (CMT) are not designed for high-capacity and large-latency networks; they often have performance problems transferring large data files over shared long-distance wide area networks. It has been shown that SCTP-CMT is more sensitive to receive buffer (rbuf) constraints, and this rbuf-blocking problem causes considerable throughput loss when multiple paths are used simultaneously. In this research paper, we demonstrate the weakness of SCTP-CMT rbuf constraints, and we then identify that rbuf-blocking problem in SCTP multihoming is mostly due to its loss-based nature for detecting network congestion. We present a simulation-based performance comparison of FAST TCP versus SCTP in high-speed networks for solving a number of throughput issues. This work proposes an end-to-end transport layer protocol (i.e., FAST TCP multihoming as a reliable, delaybased, multihome-aware, and selective ACK-based transport protocol), which can transfer data between a multihomed source and destination hosts through multiple paths simultaneously. Through extensive ns-2 simulations, we show that FAST TCP multihoming achieves the desired goals under a variety of network conditions. The experimental results and survey presented in this research also provide an insight on design decisions for the future high-speed multihomed transport layer protocols.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.5
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• pp.1139-1145
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• 2010

Wireless mesh network is similar to ad-hoc network, so when transferred to the data packet in the wireless environment, interfered factor arise. When TCP(Transport Control Protocol) was created, however as it was design based on wired link, wireless link made more transmission error than wired link. It is existent problem that TCP unfairness and congestion collapse over wireless mesh network. But packet losses due to transmission errors are more frequent. The cause of transmission error in wireless ad-hoc network may be inexactly regarded as indications of network congestion. And then, Congestion Control Algorithm was running by this situation causes the TCP performance degradation. In this paper, proposed TCP can adaptively regulate the congestion window through moving node in the Wireless Mesh Network. And it enhanced the performance.

• Journal of Communications and Networks
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• v.9 no.3
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• pp.236-246
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• 2007

This paper investigates the performance of the transmission control protocol (TCP) transport protocol over IEEE 802.11 infrastructure based wireless networks. A wireless link is generally characterized by high transmission errors, random interference and a varying latency. The erratic packet losses usually lead to a curbing of the flow of segments on the TCP connection and thus limit TCP's performance. This paper examines the impact of the lossy nature of IEEE 802.11 wireless networks on the TCP performance and proposes a scheme to improve the performance of TCP over wireless links. A negative acknowledgment scheme, selective negative acknowledgment (SNACK), is applied on TCP over wireless networks and a series of ns-2 simulations are performed to compare its performance against that of other TCP schemes. The simulation results confirm that SNACK and its proposed enhancement SNACK-S, which incorporates a bandwidth estimation model at the sender, outperform conventional TCP implementations in 802.11 wireless networks.

• Journal of Communications and Networks
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• v.11 no.5
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• pp.518-528
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• 2009

IEEE 802.11 wireless local area network (WLAN) has become the prevailing solution for wireless Internet access while transport control protocol (TCP) is the dominant transport-layer protocol in the Internet. It is known that, in an infrastructure-based WLAN with multiple stations carrying long-lived TCP flows, the number of TCP stations that are actively contending to access the wireless channel remains very small. Hence, the aggregate TCP throughput is basically independent of the total number of TCP stations. This phenomenon is due to the closed-loop nature of TCP flow control and the bottleneck downlink (i.e., access point-to-station) transmissions in infrastructure-based WLANs. In this paper, we develop a comprehensive analytical model to study TCP dynamics in infrastructure-based 802.11 WLANs. We calculate the average number of active TCP stations and the aggregate TCP throughput using our model for given total number of TCP stations and the maximum TCP receive window size. We find out that the default minimum contention window sizes specified in the standards (i.e., 31 and 15 for 802.11b and 802.11a, respectively) are not optimal in terms of TCP throughput maximization. Via ns-2 simulation, we verify the correctness of our analytical model and study the effects of some of the simplifying assumptions employed in the model. Simulation results show that our model is reasonably accurate, particularly when the wireline delay is small and/or the packet loss rate is low.

• Journal of Communications and Networks
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• v.13 no.3
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• pp.277-285
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• 2011

Real-time multimedia streaming over the Internet is rapidly increasing with the popularity of user-created contents, Web 2.0 trends, and P2P (peer-to-peer) delivery support. While many homes today are broadband-enabled, the quality of experience (QoE) of a user is still limited due to frequent interruption of media playout. The vulnerability of TCP (transmission control protocol), the popular transport-layer protocol for streaming in practice, to the packet losses, retransmissions, and timeouts makes it hard to deliver a timely and persistent flow of packets for online multimedia contents. This paper presents TCP-real-time online multimedia environment (ROME), a novel transport-layer framework that allows the establishment and coordination of multiple many-to-one TCP connections. Between one client with multiple home addresses and multiple co-located or distributed servers, TCP-ROME increases the total throughput by aggregating the resources of multiple TCP connections. It also overcomes the bandwidth fluctuations of network bottlenecks by dynamically coordinating the streams of contents from multiple servers and by adapting the streaming rate of all connections to match the bandwidth requirement of the target video.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.837-840
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• 2003

Recently the multimedia streaming traffic such as digital audio and video in the Internet has increased tremendously. Unlike TCP, the UDP protocol, which has been used to transmit streaming traffic through the Internet, does not apply any congestion control mechanism to regulate the data flow through the shared network. If this trend is let go unchecked, these traffic will effect the performance of TCP, which is used to transport data traffic, and may lead to congestion collapse of the Internet. To avoid any adverse effort on the current Internet functionality, A study on a new protocol of modification or addition of some functionality to existing transport protocol for transmitting streaming traffic in the Internet is needed. TCP-frienly congestion control mechanism is classified with window-based congestion control scheme and rate-based congestion control scheme. In this paper, we propose an algorithm for improving the transmitting rate on a hybrid TCP-friendly congestion control scheme combined with widow-based and rate-based congestion control for multimedia streaming in the internet.

• Journal of IKEEE
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• v.11 no.4
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• pp.331-339
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• 2007

The problem of architecting a reliable transport system across an overlay network using split TCP connections as the transport primitive is mainly considered. The considered overlay network uses the application-level switch in each intermediate host. We first argue that natural designs based on store-and-forward principles that are maintained by split TCP connections of hop-by-hop approaches. These approaches in overlay networks do not concern end-to-end TCP semantics. Then, a new transport protocol-Overlay Transport Protocol (OTP)-that manages the end-to-end connection and is responsible for the congestion/flow control between source host and destination host is proposed. The proposed network model for the congestion and flow control mechanisms uses a new window size-Ownd-and a new timer in the source host and destination host. We validate our analytical findings and evaluate the performance of our OTP using a prototype implementation via simulation.

• Proceedings of the Korean Information Science Society Conference
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• 2005.07a
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• pp.334-336
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• 2005

최근 오디오나 비디오 스트리밍과 같은 멀티미디어 트래픽이 증가하고 있다. 이러한 트래픽들은 패킷을 전달하는데 대부분 UDP(User Datagram Protocol)기반의 RTP(Realtime Transport Protocol)를 사용한다. 하지만 UDP기반의 RTP는 기본적으로 혼잡제어 메커니즘이 없으며 현재 인터넷의 주요 트래픽인 TCP(Transmission Control Protocol)와의 형평성을 보장하지 않는다는 문제점을 갖는다. 본 논문에서는 스트리밍 트래픽의 TCP 친화적인 전송률 조절 기법으로 TF-RTP(TCP-Friendly RTP)를 제안하였다. TF-RTP는 네트워크 상태가 혼잡하여 패킷 손실이 발생할 경우, 개선된 파라미터들을 사용하여 경쟁하는 TCP의 전송률을 보다 정확하게 계산하여 스트리밍 트래픽의 전송률을 조절함으로써 경쟁하는 TCP 트래픽과 친화적으로 동작하며 네트워크 대역폭을 보다 공평하게 사용하게 된다. 실험을 통해 제안한 TF-RTP가 TCP의 전송률을 보다 정확하게 계산하며 TCP 친화성, 공평성 측면에서도 성능 개선을 보임을 확인할 수 있었다.