With the approval of IEEE 1901 standard for power line communications (PLC) and the recent Internet-enable home appliances like the IPTV having access to a content-on-demand service through the Internet as AcTVila in Japan, there is no doubt that PLC has taken a great step forward to emerge as the preeminent in-home-network technology. However, existing schemes developed so far have not considered the PLC network connected to an unstable Internet environment (i.e. more realistic situation). In this paper, we investigate the communication performance from the end-user's perspective in networks with large and variable round-trip time (RTT) and with the existence of cross-traffic. Then, we address the problem of unfair bandwidth allocation when multiple and different types of flows coexist and propose a TCP rate control considering the difference in terms of end-to-end delay to solve it. We validate our methodology through simulations, and show that it effectively deals with the throughput unfairness problem under critical communication environment, where multiple flows with different RTTs share the PLC and cross-traffic exists on the path of the Internet.

To raise the energy efficiency of wireless clients, it is important to sleep in idle periods. When multiple network applications are running concurrently on a single wireless client, packets of each application are sent and received independently, but multiplexed at MAC-level. This uncoordinated behavior makes it difficult to control of sleep timing. In addition, frequent state transitions between active and sleep modes consume non-negligible energy. In this paper, we propose a transport-layer approach that resolves this problem and so reduces energy consumed by multiple TCP flows on a wireless LAN (WLAN) client. The proposed method, called SCTP tunneling, has two key features: flow aggregation and burst transmission. It aggregates multiple TCP flows into a single SCTP association between a wireless client and an access point to control packet transmission and reception timing. Furthermore, to improve the sleep efficiency, SCTP tunneling reduces the number of state transitions by handling multiple packets in a bursty fashion. In this study, we construct a mathematical model of the energy consumed by SCTP tunneling to assess its energy efficiency. Through numerical examples, we show that the proposed method can reduce energy consumption by up to 69%.

In order to prevent the degradation of TCP performance while traversing two WLANs, we present an implementation design of an inter-domain TCP handover method based on cross-layer and multi-homing. The proposed handover manager (HM) in the transport layer uses two TCP connections previously established via two WLANs (multi-homing) and switches the communication path between the two connections according to the handover trigger and the comparison of new/old APs. The handover trigger and comparison are conducted by assessing the wireless link quality using the frame-retry information obtained from the MAC layer (cross-layer). In a previous study, we proposed a preliminary concept for this method and evaluated its functional effectiveness through simulations. In the present study, we design an implementation considering a real system and then examine the effective performance in a real environment because a real system has several system constraints and suffers from fluctuations in an actual wireless environment. Indeed, depending on the cross-layer design, the implementation often degrades the system performance even if the method exhibits good functional performance. Moreover, the simple assessments of wireless link quality in the previous study indicated unnecessary handovers and inappropriate AP selection in a real environment. Therefore, we herein propose a new architecture that performs cross-layer collaboration between the MAC layer and the transport layer while avoiding degradation of system performance. In addition, we use a new assessment scheme of wireless link quality, i.e., double thresholds of frame retry and comparison of frame retry ratio, in order to prevent handover oscillation caused by fluctuations in the wireless environment. The experimental results demonstrate that the prototype system works well by controlling two TCP connections based on assessments of wireless link quality thereby achieving efficient inter-domain TCP handover in a real WLAN environment.

Estimating the packet loss ratio of TCP transfers is essential for passively measuring Quality of Service (QoS) on the Internet traffic. However, only a few studies have been conducted on this issue. The Benko-Veres algorithm is one technique for estimating the packet loss ratio of two networks separated by a measurement point. However, this study shows that it leads to an estimation error of a few hundred percent in the particular environment where the packet loss probabilities between the two networks are asymmetrical. We propose a passive method for packet loss estimation that offers improved estimation accuracy by introducing classification conditions for the TCP retransmission timeout. An experiment shows that our proposed algorithm suppressed the maximum estimation error to less than 15%.

A packet memory stores packets in internet routers and it requires typically RTTC for the buffer space, e.g. several GBytes, where RTT is an average round-trip time of a TCP flow and C is the bandwidth of the router's output link. It is implemented with DRAM parts which are accessed in parallel to achieve required bandwidth. They consume significant power in a router whose scalability is heavily limited by power and heat problems. Previous work shows the packet memory size can be reduced to , where N is the number of long-lived TCP flows. In this paper, we propose a novel packet memory architecture which splits the packet memory into on-chip and off-chip packet memories. We also propose a low-power packet mapping method for this architecture by estimating the latency of packets and mapping packets with small latencies to the on-chip memory. The experimental results show that our proposed architecture and mapping method reduce the dynamic power consumption of the off-chip memory by as much as 94.1% with only 50% of the packet buffer size suggested by the previous work in realistic scenarios.

This letter proposes a new mechanism that supports adaptive sending control using Real-Time Streaming Protocol (RTSP) and Transmission Control Protocol (TCP) for IPTV service over heterogeneous networks. The proposed mechanism is implemented on a mobile IPTV device and its performance is verified for providing seamless television watching in heterogeneous networks, even when in motion.

We address the global asymptotic stability of FAST TCP, especially considering cross traffics, time-varying network feedback delay, and queuing delay dynamics at link. Exploiting the inherent dynamic property of FAST TCP, we construct two sequences that represent the lower and upper bound variations of the congestion window in time. By showing that the sequences converge to the equilibrium point of the congestion window, we establish that FAST TCP in itself is globally asymptotically stable without any specific conditions on the tuning parameter α or the update gain γ.

We analyze measured traffic data to investigate the characteristics of TCP quality metrics such as packet retransmission rate, roundtrip time (RTT), and throughput of connections classified by their type (client-server (C/S) or peer-to-peer (P2P)), or by the location of the connection host (domestic or overseas). Our findings are as follows. (i) The TCP quality metrics of the measured traffic data are not necessarily consistent with a theoretical formula proposed in a previous study. However, the average RTT and retransmission rate are negatively correlated with the throughput, which is similar to this formula. Furthermore, the maximum idle time, which is defined as the maximum length of the packet interarrival times, is negatively correlated with throughput. (ii) Each TCP quality metric of C/S connections is higher than that of P2P connections. Here “higher quality” means that either the throughput is higher, or the other TCP quality metrics lead to higher throughput; for example the average RTT is lower or the retransmission rate is lower. Specifically, the median throughput of C/S connections is 2.5 times higher than that of P2P connections in the incoming direction of domestic traffic. (iii) The characteristics of TCP quality metrics depend on the location of the host of the TCP connection. There are cases in which overseas servers might use a different TCP congestion control scheme. Even if we eliminate these servers, there is still a difference in the degree of impact the average RTT has on the throughput between domestic and overseas traffic. One reason for this is thought to be the difference in the maximum idle time, and another is the fact that congestion levels of these types of traffic differ, even if their average RTTs are the same.

Some researchers have started using the high-speed communication satellite WINDS to evaluate TCP congestion control methods. However, they do not aim to maximize TCP throughput when WINDS is paired with a Large Earth Station (LET). This paper evaluates the typical TCP performance of satellite links over a network composed of LET and WINDS in order to break the TCP throughput record for satellite links.

Internet behavior is becoming more complex due to ever-changing networking technologies and applications. Thus, understanding and controlling the complex behavior of the Internet are important for designing future networks. One of the complex behaviors of the Internet is traffic dynamics. Previous studies revealed that flow control in the transport layer affects the traffic dynamics of the Internet. However, it is not clear how the topological structure impacts traffic dynamics. In this paper, we investigate packet delay dynamics and traffic fluctuation in ISP router-level topologies where the degree distribution exhibits a power-law nature, and the nodes interact via end-to-end feedback control functionality. We show the packet delay dynamics of the BA topologies generated by the Barabasi-Albert (BA) model and the ISP router-level topologies. Simulation results show that the end-to-end delay distributions exhibit a heavy tail in the TCP model. Moreover, the number of links with highly fluctuating queue length increases dramatically compared to that in the stop-and-wait model. Even in this case, the high-modularity structures of the ISP topologies reduce the number of highly fluctuating links compared with the BA topologies.

Cubic TCP, one of transport protocols designed for high bandwidth-delay product (BDP) networks, has successfully been deployed in the Internet. Multi-homed computers with multiple interfaces to access the Internet via high speed links will become more popular. In this work, we introduce an extended version of Cubic TCP for multiple paths, called MPCubic. The extension process is approached from an analysis model of Cubic by using coordinated congestion control between paths. MPCubic can spread its traffic across paths in load-balancing manner, while preserving fair sharing with regular TCP, Cubic, and MPTCP at common bottlenecks. Moreover, to improve resilience to link failure, we propose a multipath fast recovery algorithm. The algorithm can significantly reduce the recovery time of data rate after restoration of failed links. These techniques can be useful for resilient high-bandwidth applications (for example, tele-health conference) in disaster-affected areas. Our simulation results show that MPCubic can achieve stability, throughput improvement, fairness, load-balancing, and quick data rate recovery from link failure under a variety of network conditions.

To improve TCP throughput even if the maximum receiving window size is small, a TCP performance enhancing proxy (PEP) using a UDP-like packet sending policy with error control has been proposed. The PEP operates on a router along a TCP connection. When the PEP receives a data packet from the source host, it transmits the packet to the destination host, copies the packet into the local buffer (PEP buffer) in case the packets need to be transmitted and sends a premature ACK acknowledging receipt of the packet to the source host. In the PEP, the number of prematurely acknowledged packets in the PEP buffer is limited to a fixed threshold (watermark) value to avoid network congestion. Although the watermark value should be adjusted to changes in the network conditions, watermark adjusting algorithms have not been investigated. In this paper, we propose a watermark adjusting algorithm the goal of which is to maximize the throughput of each connection as much as possible without excessively suppressing the throughputs of the other connections. In our proposed algorithm, a newly established connection uses the initial watermark value of zero to avoid drastic network congestion and increases the value as long as its throughput increases. In addition, when a new connection is established, every already-established connection halves its watermark value to allow the newly established connection to use some portion of the bandwidth and increases again as long as its throughput increases. We compare the proposed algorithm (CW method) with other methods: the FW method that uses a fixed large watermark value and the NP method that does not use the PEP. Numerical results with respect to throughput and fairness showed that the CW method is generally superior to the other two methods.

A transmission control protocol (TCP) using an additive increase multiplicative decrease (AIMD) algorithm for congestion control plays a leading role in advanced Internet services. However, the AIMD method shows only low link utilization in lossy networks with long delay such as satellite networks. This is because the cwnd dynamics of TCP are reduced by long propagation delay, and TCP uses an inadequate congestion control algorithm, which does not distinguish packet loss from wireless errors from that due to congestion of the wireless networks. To overcome these problems, we propose an exponential recovery (ER) TCP that uses the exponential recovery function for rapidly occupying available bandwidth during a congestion avoidance period, and an adaptive congestion window decrease scheme using timestamp base available bandwidth estimation (TABE) to cope with wireless channel errors. We simulate the proposed ER-TCP under various test scenarios using the ns-2 network simulator to verify its performance enhancement. Simulation results show that the proposal is a more suitable TCP than the several TCP variants under long delay and heavy loss probability environments of satellite networks.

TCP Reno is not fully utilized in under-buffered links. We propose a new TCP congestion control algorithm that can utilize the link almost up to 100% except the first congestion avoidance cycle. Our scheme estimates the minimum congestion window size for full link utilization in every congestion avoidance cycle and sends extra packets without touching TCP Reno congestion control. It has the same RTT fairness and the same saw-tooth wave as TCP Reno does. Our scheme does not affect competing TCP Reno flows since it uses only unused link capacity. We provide a simple mathematical modeling as well as ns-2 simulation results which show that the link utilization is improved by up to 19.88% for k=1/8 against TCP Reno when the buffer is k times the optimal buffer size. We claim that our scheme is useful for transmitting large amount of data in under-buffered links.

The explosive growth of Internet usage has caused problems for the current Internet in terms of traffic congestion within networks and performance degradation of end-to-end flows. Therefore, a reconsideration of the current Internet has begun and is being actively discussed worldwide with the goals of enabling efficient share of limited network resources (i.e., the link bandwidth) and improved performance. To directly address the inefficiency of TCP's congestion mitigation solely on the end-to-end basis, in this paper we propose an adaptive split connection scheme on advanced relay nodes; this scheme dynamically splits end-to-end TCP connections on the basis of congestion status in output links. Through simulation evaluations, we examine the effectiveness and potential of the proposed scheme.

TCP Friendly Rate Control (TFRC) has been widely used in the Internet multimedia streaming applications. However, performance of traditional TFRC algorithm degrades significantly when deployed over wireless networks. Although numerous TFRC variants have been proposed to improve the performance of TFRC over wireless networks, designing a TFRC algorithm with graceful performance both in throughput and fairness still remains a great challenge. In this paper, a novel TFRC algorithm, named TFRC-FIT, is proposed to improve the performance of TFRC over wireless environments. In the proposed approach, the behavior of multiple TFRC flows is simulated in single connection, while the number of simulated flows is adjusted by the network queuing delay. Through this mechanism, TFRC-FIT can fully utilize the capacity of wireless networks, while maintaining good fairness and TCP friendliness. Both theoretical analysis and extensive experiments over hardware network emulator, Planetlab test bed as well as commercial 3G wireless networks are carried out to characterize and validate the performance of our proposed approach.

Packet pacing is a well-known technique for reducing the short-time-scale burstiness of traffic, and software-based packet pacing has been categorized into two approaches: the timer interrupt-based approach and the gap packet-based approach. The former was originally hard to implement for Gigabit class networks because it requires the operating system to handle too frequent periodic timer interrupts, thus incurring a large overhead. On the other hand, a gap packet-based packet pacing mechanism achieves precise pacing without depending on the timer resolution. However, in order to guarantee the accuracy of rate control, the system must be able to transmit packets at the wire rate. In this paper, we propose a high-resolution timer-based packet pacing mechanism that determines the transmission timing of packets by using a sub-microsecond resolution timer. The high-resolution timer is a light-weight mechanism compared to the traditional low-resolution periodic timer. With recent progress in hardware protocol offload technologies and multicore-aware network protocol stacks, we believe high-resolution timer-based packet pacing has become practical. Our experimental results show that the proposed mechanism can work on a wider range of systems without degrading the accuracy of rate control. However, a higher CPU load is observed when the number of traffic classes increases, compared to a gap packet-based pacing mechanism.

This paper proposes a resource allocation method based on TCP (Transmission Control Protocol) throughput for base station diversity systems. A goal of this study is to achieve high throughput wireless Internet access by utilizing multiple wireless links effectively. The conventional work showed that base station diversity techniques can improve TCP performance. However, the improvement depends on the wireless environment of the wireless terminal. The proposed resource allocation method allocates wireless links to a wireless terminal based on its estimated TCP throughput and current traffic of each base station. Our method can take account of some network protocols such as TCP and UDP (User Datagram Protocol) by measuring the current traffic of each base station. In addition, wireless links are preferentially assigned to the wireless terminal that has the largest performance improvement per wireless link. Therefore, the proposal provides better overall system performance than the previous technique.

We address the stability property of the FAST TCP congestion control algorithm. Based on a continuous-time dynamic model of the FAST TCP network, we establish that FAST TCP in itself is globally exponentially stable without any specific conditions on the congestion control parameter or the update gain. Simulation results demonstrate the validity of the global exponential stability of FAST TCP.

Mechanisms to provide QoS for IP communication have been frequently discussed recently. For example, in ITU-T and ETSI, the Next Generation Network has been discussed, and there are many reports on providing QoS for real-time services using RTP. However, in the current Internet, Transmission Control Protocol (TCP) is a major transport-layer protocol, and many real-time services are using TCP. In this paper, we present a performance evaluation of TCP under the control of a token bucket policer, which is one of the most common policing functions, and derive a formula of TCP throughput. We also evaluate the accuracy of our model by comparing results of simulations and experiments.