Recently, real-time media delivery services such as video streaming and VoIP have rapidly become popular. For these applications requiring high-level QoS guarantee, our research group has proposed a transport-layer approach to provide predictable throughput for upper-layer applications. In the present paper, we propose a congestion control mechanism of TCP for achieving predictable throughput. It does not mean we can guarantee the throughput, while we can provide the throughput required by an upper-layer application at high probability when network congestion level is not so high by using the inline network measurement technique for available bandwidth of the network path. We present the evaluation results for the proposed mechanism obtained in simulation and implementation experiments, and confirm that the proposed mechanism can assure a TCP throughput if the required bandwidth is not so high compared to the physical bandwidth, even when other ordinary TCP (e.g., TCP Reno) connections occupy the link.

With the rapid development of wireless network technologies, heterogeneous networks with wired and wireless links are becoming common. However, the performance of TCP data transmission deteriorates significantly when a TCP connection traverses such networks, mainly because of packet losses caused by the high bit error rate of wireless links. Many solutions for this problem have been proposed in the past literature. However, most of them have various drawbacks, such as difficulties in their deployment by the wireless access network provider and end users, violation of TCP's end-to-end principle by splitting the TCP connection, or inapplicability to IP-level encrypted traffic because the base station needs to access the TCP header. In this paper, we propose a new mechanism without such drawbacks to improve the performance of TCP over wired and wireless heterogeneous networks. Our mechanism employs a receiver-based approach, which does not need modifications to be made to the sender TCP or the base station. It uses the ACK-splitting method for increasing the congestion window size quickly in order to restrain the throughput degradation caused by packet losses due to the high bit error rate of wireless links. We evaluate the performance of our mechanism and show that our mechanism can increase throughput by up to 94% in a UMTS network. The simulation results also show that our mechanism does not significantly deteriorate even when the receiver cannot perfectly distinguish whether packet losses are due to network congestion or bit errors on the wireless links.

TCP performance in the IEEE 802.11-based multihop ad hoc networks is extremely poor, because the congestion control mechanism of TCP cannot effectively deal with the problem of packet drops caused by mobility and shared channel contention among wireless nodes. In this paper, we present a cross-layer method, which adaptively adjusts the TCP maximum window size according to the number of RTS (Request To Send) retry counts of the MAC layer at the TCP sender, to control the number of TCP packets in the network and thus decrease the channel contention. Our simulation results show that this method can remarkably improve TCP throughput and its stability.