• Journal of KIISE:Computing Practices and Letters
• /
• v.15 no.2
• /
• pp.114-124
• /
• 2009

In this paper, we propose a practical TCP friendly rate control scheme that considers the minimum channel bandwidth of the network when transporting SVC (scalable video coding) video over IP netowrks such as Internet. RTP and RTCP is mainly designed for use with UDP (User Datagram Protocol) for real-time video transport over the Internet. TCP-friendly rate control was proposed to satisfy the demands of multimedia applications while being reasonably fair when competing for bandwidth with conventional TCP applications. However the rate control model of the conventional TCP-friendly rate control scheme does not consider the minimum channel bandwidth of the network. Thus the estimated channel bandwidth by the conventional rate control model might be quite different from the real channel bandwidth when the packet loss ratio of the network is very large. In this paper, we propose a modified TCP-friendly rate control scheme that considers the minimum channel bandwidth of the network. Based on the modified TCP-friendly rate control, we assign the minimum channel bandwidth to the base layer bitstream of SVC video, and remaining available bandwidth is allocated to the enhancement layer of SVC video for the TCP friendly scalable video transmission. It is shown by simulations that the modified TCP-friendly rate control scheme can be effectively used for a wider range of controlled bit rates depending on the packet loss ratio than the conventional TCP-friendly control scheme. Furthermore, the effectiveness of the proposed scheme in terms of objective video quality is proved by comparing PSNR performance with the conventional scheme.

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

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

• Proceedings of the Korean Information Science Society Conference
• /
• 2005.11a
• /
• pp.154-156
• /
• 2005

본 논문은 TCP의 무선 환경에서의 전송 성능 감소와 패킷 왕복 시간에 따른 대역폭 선점 문제를 해결하기 위한 타이머 기반의 혼잡 제어 방식을 제안한다. TCP는 패킷 손실 확률이 많은 무선 환경에서 네트워크 혼잡에 의한 패킷 손실을 방지하기 위한 느린 전송률 증가로 인해 전송 성능이 크게 떨어진다. 또한, TCP 송신자는 전송률을 결정하는 전송 윈도우를 수신자로부터 응답 메시지를 받을 때만 조정시키므로, 패킷의 왕복 시간의 차이에 따른 전송률 편중 현상과 다수개의 응답 메시지에 의한 과도한 트래픽 발생의 문제가 발생한다. 본 논문에서 제안하는 타이머 기반의 TCP 혼잡 제어 방식은 패킷의 전송 시간 간격을 타이머로 조정함으로써 무선 환경에서 전송 성능을 향상시키고 패킷의 왕복 시간 차이에 따른 전송률 편중 현상을 완화시키며 다수개의 응답 메시지에 의한 과도한 트래픽의 발생을 제한한다. 제안하는 방법은 실제 환경에서 구현되었으며, 다양한 네트워크 환경에서의 실험을 통해 그 성능을 입증하였다.

• Proceedings of the Korean Information Science Society Conference
• /
• 2004.10c
• /
• pp.181-183
• /
• 2004

본 논문은 TCP에서 사용되는 혼잡 제어 방식의 문제점인 전송 시간에 대한 전송률 편중 현상을 완화시키기 위한 비례 증가 혼잡 제어 방식을 제안한다. TCP는 네트워크의 혼잡 상황을 제어하기 위하여 ACK 기반의 혼잡 윈도우 조정 방식을 사용한다. 그러나 기존 TCP에서 사용되는 혼잡 제어 방식은 그 자체적으로 네트워크의 혼잡 상황을 유도하며, 전송 시간이 다른 두 개의 TCP 연결이 하나의 병목 지점을 공유할 경우, 전송 시간의 차이에 따라 전송률의 편중 현상이 나타난다. 본 논문에서는 이러한 네트워크 혼잡 상황 유발과 전송 시간의 차이에 따른 전송률 편중 현상을 완화시키기 위한 송신자 기반의 네트워크 대역폭 측정 방법 및 비례 증가 혼잡 제어 알고리즘을 제안하며, 구현 및 실험을 통해 기존 TCP의 문제점을 해결할 수 있음을 보인다.

• 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.

• KIPS Transactions on Computer and Communication Systems
• /
• v.2 no.3
• /
• pp.117-124
• /
• 2013

In this paper we propose a TFRC(TCP Friendly Rate Control) which guarantees a minimum rate in order to improve the efficiency of the previous TFRC which cannot distinguish congestion losses and wireless losses and decreases throughput both in wired and wireless networks. This TFRC technique is able to guarantee a minimum rate for video by restricting a loss event rate with packet loss probability about existing TFRC and constraining a rate reduction from the feedback timeout. When we experimented both the existing TFRC and the new one with TCP in the same network, we found that the latter is better than the former. Consequently, it shows that the proposed TFRC can improve video streaming quality using a guaranteed minimum transmission rate.

• Journal of KIISE:Information Networking
• /
• v.34 no.6
• /
• pp.505-513
• /
• 2007

Vertical handoff is a new type of handoff in wireless networks. It is issued when a mobile node moves over overlapping wireless networks with each proving a different access bandwidth, transmission latency, and coverage. By issuing the vertical handoff, the mobile node can obtain better network bandwidth. In the sense of TCP throughput, however, the vertical handoff does not always produce positive performance gain, so sometimes it is better for the mobile node to stay at lower bandwidth providing network rather than to select and move to higher bandwidth providing network. In this paper, we analyze TCP throughput for vertical handoff, and propose a new handoff decision scheme which can estimate TCP throughput at the moment of vertical handoff. Based on the estimation, a mobile node can decide to issue vertical handoff to produce better TCP throughput, and we verify the results by simulations.

• The KIPS Transactions:PartC
• /
• v.12C no.3 s.99
• /
• pp.395-400
• /
• 2005

The IEEE 802.11b WLAN supports multiple transmission rates and the rate is chosen in an adaptive manner by an auto rate control algorithm. This auto rate control algorithm deeply affects the total system performance of the IEEE 802.11b WLAN. In this paper, we examine the WLAN performance with regard to the auto rate control algorithm especially the ARF scheme. The experimental results indicate that the ARF scheme works well in the face of signal noise due to node location. However, the ARF scheme severely degrades system performance when multiple nodes contend to obtain the wireless channel and the packet is lost due to signal collision. In addition, TCP prevent the performance degradation due to ARF scheme by retaining number of active nodes. However, some applications, such as transporting multimedia data, adopt the UDP. Therefore, the TCP cannot be an optimal solution for all WLAN applications.

• Journal of Advanced Navigation Technology
• /
• v.12 no.2
• /
• pp.154-163
• /
• 2008

In a mobile broadband wireless access (MBWA) network, many users access a base station (BS), which relays data transferred from high-speed wired network to low-speed wireless network. For this difference of their data rate, a BS suffers from the lack of its buffer space when many users run multiple applications at the same time, and thus packet losses occur. TCP, which guarantees end-to-end reliability, is used as transport protocol also in wireless networks. But TCP lowers their transmission rate incorrectly and frequently whenever packet losses occur. And they increase their transmission rate differently with each other; finally TCP throughput of each TCP flow varies largely, and then TCP fairness goes worse. In this paper, a scheme that controls packet transmission rate adaptively according to TCP flows' transmission rate, that prevents buffer overflows at BS, and that guarantees TCP fairness at a certain degree is proposed. As it is analyzed by simulations, the proposed scheme enhances TCP fairness by maintaining TCP throughput of each TCP sender similarly with each other.