Streaming services that use TCP have increased; however, throughput is unstable due to congestion control caused by packet loss when TCP is used. Thus, TCP control to secure a required transmission rate for streaming communication using Forward Error Correction (FEC) technology (TCP-AFEC) has been proposed. TCP-AFEC can control the appropriate transmission rate according to network conditions using a combination of TCP congestion control and FEC. However, TCP-AFEC was not developed for wireless Local Area Network (LAN) environments; thus, it requires a certain time to set the appropriate redundancy and cannot obtain the required throughput. In this paper, we demonstrate the drawbacks of TCP-AFEC in wireless LAN environments. Then, we propose a redundancy setting method that can secure the required throughput for FEC, i.e., TCP-TFEC. Finally, we show that TCP-TFEC can secure more stable throughput than TCP-AFEC.

We analyze the defects of window-based TCP algorithm in datacenter networks and propose Rate-based Datacenter TCP (RDT) algorithm in this paper. The RDT algorithm combines rate-based congestion control technology with ECN (Explicit Congestion Notification) mechanism of DCTCP. The experiments in NS2 show that RDT has a potential to completely avoid TCP incast collapse in datacenters and inherit the low latency advantages of DCTCP.

Two congestion control schemes designed specifically to handle changes in the datalink interface of a mobile host are presented. The future mobile environment is expected to involve multimode connectivity to the Internet and dynamic switching of the connection mode depending on network conditions. The conventional Transmission Control Protocol (TCP), however, is unable to maintain stable and efficient throughput across such interface changes. The two main issues are the handling of the change in host Internet Protocol (IP) address, and the reliability and continuity of TCP flow when the datalink interface changes. Although existing architectures addressing the first issue have already been proposed, the problem of congestion control remains. In this paper, considering a large change in bandwidth when the datalink interface changes, two new schemes to address these issues are proposed. The first scheme, Immediate Expiration of Timeout Timer, detects interface changes and begins retransmission immediately without waiting for a retransmission timeout as in existing architectures. The second scheme, Bandwidth-Aware Slow Start Threshold, detects the interface change and estimates the new bandwidth so as to set an appropriate slow start threshold for retransmission. Through simulations, the proposed schemes are demonstrated to provide marked improvements in performance over existing architectures.

Satellite Internet is one of the most important networks for emergency communications because of its tolerant of disasters such as earthquake. Therefore, satellite Internet has received considerable attention over recent years. However, most standard implementations of TCP congestion control method perform poorly in satellite Internet due to its high bit error rate and long propagation delay. This paper proposes a new TCP congestion control method called TCP-STAR to improve the throughput over satellite Internet. TCP-STAR has three new mechanisms, namely Congestion Window Setting (CWS) based on available bandwidth, Lift Window Control (LWC), and Acknowledgment Error Notification (AEN). CWS can resist the reduction of the transmission rate when data losses are caused by bit error. LWC is able to increase the congestion window quickly based on the estimated available bandwidth. AEN can avoid the reduction of the throughput by mis-retransmission of data. The mis-retransmission is caused by ack losses or delay. Simulations show that TCP-STAR can obtain the best throughput comparing with other TCP variants (TCP-J and TCP-WestwoodBR). Furthermore, we found that the fairness of TCP-STAR is a little lower than that of TCP-WestwoodBR. However, the fairness of TCP-STAR is equal to TCP-J.

TCP congestion control is receiving increased attention in recent years due to their usefulness for network stability, robustness use of network buffer and bandwidth resources on an end-to-end per-connection basis. The RED scheme was designed for a network where a single dropped packet is sufficient to signal the presence of congestion to the TCP protocol. This paper applies matrix-analytic approach to analyze both the long-term and the short-term drop behaviors of a queue with RED scheme and uses this model to quantify the benefits brought about by RED. The result shows that the drop probability between RED and Drop-Tail is very close under heavy load conditions. This indicates that RED not only can resolve the synchronization problem but also has the same loss performance with Drop-Tail scheme under the heavy load circumstances. Our findings also show that the rate oscillation behavior of RED is better than Drop-Tail when TCP applies the additive-increase and multiplication-decrease mechanism. As a consequence, it can help reduce the required buffer capacity in the RED router.