• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.45 no.9
• /
• pp.30-38
• /
• 2008

The traditional TCP was designed assuming wired networks. Thus, if it is used networks consisting of both wired and wireless networks, all packet losses including random losses in wireless links are regarded as network congestion losses. Misclassification of packet losses causes unnecessary reduction of transmission rate, and results in waste of bandwidth. In this paper, we present WTCP(wireless TCP) congestion control algorithm that differentiates the random losses more accurately, and adopts improved congestion control which results in better network throughput. To evaluate the performance of proposed scheme, we compared the proposed algorithm with TCP Westwood and TCP Veno via simulations.

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

• Proceedings of the Korean Information Science Society Conference
• /
• 2012.06d
• /
• pp.291-293
• /
• 2012

급격히 증가하는 모바일 디바이스에 의한 모바일 데이터 요구량을 처리하기 위해 가장많이 고려되고 있는 IEEE 802.11의 WLAN은 셀룰러보다 높은 데이터 전송속도를 제공하지만 이동성이 고려되어있지 않다. 이에 본 논문에서는 Enterprise WLAN에서 IEEE 802.21 MIH 표준에 기반하여 Media Independent Information Server (MIIS)로부터 수신된 주변 AP들의 네트워크의 정보를 바탕으로 후보 엑세스 AP 네트워크를 선정 및 WiMAX에 make-before-break 핸드오버하는 방안을 제안한다. 제안하는 방안은 실제 Enterprise WLAN와 Mobile WiMAX 환경에서 이동하는 단말의 시간에 따른 TCP throughput에 대한 성능평가를 하였다. 이를 통해 제안하는 방안의 평균TCP throughput 성능이 9.04Mbits/s으로 기존 방안의 TCP throughput 성능 6.49Mbits/s보다 약 40%향상됨을 확인하였다.

• Journal of KIISE:Information Networking
• /
• v.31 no.2
• /
• pp.188-195
• /
• 2004

We have analyzed the performance of TCP in the CDMA mobile communications systems with the adaptive modulation and coding(AMC). The wireless channel using AMC is characterized with not high error rate but highly varying bandwidth. Due to time-varying bandwidth, timeout events of TCP occurs more frequently, which leads to the throughput degradation. The analysis model is composed of the two parts. In the first part, we divide TCP packet stream into ‘packet groups’and derive the probability distribution of the wireless transmission time of each Packet group that reflects the time varying characteristics of AMC. In the second part, we formulate embedded Markov chain by making use of the results of the first part to model TCP timer mechanism and wireless transmission. Since our system model is characterized by the forward link high speed data transmission using AMC, the results reported in this paper can be used as a guideline for the design and operation of HSDPA, 1xEV-DO, and 1xEV-DV.

• Journal of Communications and Networks
• /
• v.12 no.3
• /
• pp.275-284
• /
• 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 the Korea Contents Association
• /
• v.11 no.2
• /
• pp.69-78
• /
• 2011

Recently, the bandwidth available to an end user has been dramatically increasing with the advancing of network technologies. This high-speed network naturally requires faster and/or stable data transmission techniques. The UDT(UDP based Data Transfer protocol) is a UDP based transport protocol, and shows more efficient throughput than TCP in the long RTT environment, with benefit of rate control for a SYN time. With a NAK event, however, it is difficult to expect an optimum performance due to the increase of fixed sendInterval and the flow control based on the previous RTT. This paper proposes a rate control method on following a NAK, by adjusting the sendInterval according to some degree of RTT period which calculated from a set of experimental results. In addition, it suggests an improved flow control method based on the TCP vegas, in order to predict the network congestion afterward. An experimental results show that the revised flow control method improves UDT's throughput about 20Mbps. With combining the rate control and flow control proposed, the UDT throughput can be improved up to 26Mbps in average.

• Journal of KIISE:Information Networking
• /
• v.36 no.3
• /
• pp.191-197
• /
• 2009

One of popular applications in the Internet is the multimedia streaming services such as IPTV or You Tube in which supporting quality of service (QoS) is an important issue. One of widely adopted rate control scheme is TCP-friendly rate control (TFRC) which shows comparable performance with TCP in term of throughput and lower variation of throughput over time. On the other hand, devices with multiple interfaces are emerging in the market. However, it has not proposed to exploit multiple interfaces simultaneously for multimedia streaming services with TFRC. In this paper, we propose a multimedia streaming algorithm with TFRC which exploits multiple interfaces to guarantee the quality of service. We show that the proposed scheme shows better performance than those with a single interface in terms of throughput and communication costs.

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

• IEMEK Journal of Embedded Systems and Applications
• /
• v.1 no.2
• /
• pp.64-72
• /
• 2006

The long propagation delay over satellite internet causes degradation of TCP performance in slow start phase. Especially, web traffic performance is greatly reduced by low throughput in slow start phase. To improve web traffic performance, we propose the Dynamic ACK Generation Scheme which generates ACKs and considers sender RTO in PEP (Performance Enhancing Proxy). The Normal ACK generation mechanism improves TCP throughput, and also decreases sender RTO. if PEP stops generating ACKs, TCP performance will be reduced by frequent RTO expiration. To solve this problem, our scheme adjusts RTO using ACK generation interval. And it supports retransmission mechanism for loss recovery in PEP. The results of the performance analysis provide a good evidence to demonstrate the efficiency of our mechanisms over satellite internet.

• Journal of KIISE:Information Networking
• /
• v.32 no.5
• /
• pp.583-592
• /
• 2005

An important factor influencing the robustness of the Internet is the end-to-end TCP congestion control. However, the congestion control scheme of TCP Reno, the most popular TCP version on the Internet, employs passive congestion indication. It makes the network congestion worse. Brakmo and Peterson proposed a congestion control algorithm, TCP Vegas, by modifying the congestion avoidance scheme of TCP Reno. Many studies indicate that Vegas is able to achieve better throughput and higher stability than Reno. But there are three unfairness problems in Vegas. These problems hinder the spread of Vegas in the current Internet. In this paper, in order to solve these unfairness problems, we propose a new congestion control algorithm called TCP NewVegas. The proposed NewVegas is able to solve these unfairness problems effectively by using the variation of the number of queued packets in a bottleneck router. To evaluate the proposed approach, we compare the performance among NewVegas, Reno and Vegas. Through the simulation, NewVegas is shown to be able to achieve throughput and better fairness than Vegas.