The Transmission Control Protocol (TCP) with Network Coding (TCP/NC) was proposed to introduce packet loss recovery ability at the sink without TCP retransmission, which is realized by proactively sending redundant combination packets encoded at the source. Although TCP/NC is expected to mitigate the goodput degradation of TCP over lossy networks, the original TCP/NC does not work well in burst loss and time-varying channels. No apparent scheme was provided to decide and change the network coding-related parameters (NC parameters) to suit the diverse and changeable loss conditions. In this paper, a solution to support TCP/NC in adapting to mentioned conditions is proposed, called TCP/NC with Loss Rate and Loss Burstiness Estimation (TCP/NCwLRLBE). Both the packet loss rate and burstiness are estimated by observing transmitted packets to adapt to burst loss channels. Appropriate NC parameters are calculated from the estimated probability of successful recoverable transmission based on a mathematical model of packet losses. Moreover, a new mechanism for coding window handling is developed to update NC parameters in the coding system promptly. The proposed scheme is implemented and validated in Network Simulator 3 with two different types of burst loss model. The results suggest the potential of TCP/NCwLRLBE to mitigate the TCP goodput degradation in both the random loss and burst loss channels with the time-varying conditions.

With the emergence of bandwidth-greedy application services, high-speed transport protocols are expected to effectively and aggressively use large amounts of bandwidth in current broadband and multimedia networks. However, when high-speed transport protocols compete with other standard TCP flows, they can occupy most of the available bandwidth leading to disruption of service. To deploy high-speed transport protocols on the Internet, such unfair situations must be improved. In this paper, therefore, we propose a method to improve fairness, called Kyushu-TCP (KTCP), which introduces a non-aggressive period in the congestion avoidance phase to give other standard TCP flows more chances of increasing their transmission rates. This method improves fairness in terms of the throughput by estimating the stably available bandwidth-delay product and adjusting its transmission rate based on this estimation. We show the effectiveness of the proposed method through simulations.

In general, Transmission Control Protocol (TCP), e.g., TCP NewReno, considers all losses to be a sign of congestion. It decreases the sending rate whenever a loss is detected. Integrating the network coding (NC) into protocol stack and making it cooperate with TCP (TCP/NC) would provide the benefit of masking packet losses in lossy networks, e.g., wireless networks. TCP/NC complements the packet loss recovery capability without retransmission at a sink by sending the redundant combination packets which are encoded at the source. However, TCP/NC is less effective under heavy and bursty loss which often occurs in fast fading channel because the retransmission mechanism of the TCP/NC entirely relies on the TCP layer. Our solution is TCP/NC with enhanced retransmission (TCP/NCwER), for which a new retransmission mechanism is developed to retransmit more than one lost packet quickly and efficiently, to allow encoding the retransmitted packets for reducing the repeated losses, and to handle the dependent combination packets for avoiding the decoding failure. We implement and test our proposal in Network Simulator 3. The results show that TCP/NCwER overcomes the deficiencies of the original TCP/NC and improves the TCP goodput under both random loss and burst loss channels.

Differentiated Service (DiffServ) is a technology designed to provide Quality of Service (QoS) in the Internet, and is superior to Integrated Service (IntServ) technology with respect to the simplicity of its architecture and the scalability of networks. Although various simulation studies and estimations over testbeds have investigated the QoS that is offered via the DiffServ framework, almost all of them focused on the characteristics in a single DiffServ domain. However, the Internet is actually composed of a large number of AS domains, and thus packets are very likely to arrive at their destinations after going through many different domains. Therefore, we have analyzed the QoS performance in a model consisting of multiple DiffServ domains, and focused especially on the quality provided by Assured Forwarding Service (AF) to achieve statistical bandwidth allocation with AF-PHB. Our simulation results show some throughput characteristics of flows over multiple Diffserv domains, which clarify how network configurations impact the QoS over multiple DiffServ domains.

The performance of a real-time networked application can be drastically affected by delays in packets traversing the network. Some real-time applications impose limits for acceptable network delay, and so a packet which is delayed longer than the limit before arriving at its destination is worthless to the flow to which the packet belongs. Not only that, but the rejected packet is also damaging to the quality of other flows in the network, because it may increase the queuing delay for other packets. Therefore, this paper proposes an adaptive scheme using two mechanisms, in which packets experiencing too great a delay are discarded at intermediate nodes based on the delay limit for the application and the delay experienced by each packet. This earlier discarding of packets is expected to improve the overall delay performance of real-time flows competing for network resources when the network is congested. An extensive simulation is conducted, and the results show that the scheme has great potential in improving the delay performance of real-time traffic in both homogeneous and heterogeneous environments in terms of traffic volume and application delay requirements.