In this paper, we propose a packet loss detection mechanism called Interest ACKnowledgement (ACK). Interest ACK provides information on the history of successful Interest packet receptions at a repository (i.e., content provider); this information is conveyed to the corresponding entity (i.e., content consumer) via the header of Data packets. Interest ACKs enable the entity to quickly and accurately detect Interest and Data packet losses in the network. We conduct simulations to investigate the effectiveness of Interest ACKs under several scenarios. Our results show that Interest ACKs are effective for improving the adaptability and stability of CCN with window-based flow control and that packet losses at the repository can be reduced by 10%-20%. Moreover, by extending Interest ACK, we propose a lossy link detection mechanism called LLD-IA (Lossy Link Detection with Interest ACKs), which is a mechanism for an entity to estimate the link where the packet was discarded in a network. Also, we show that LLD-IA can effectively detect links where packets were discarded under moderate packet loss ratios through simulation.

This paper focuses on RTT-fairness of multiple TCP flows over the Internet, and proposes a new TCP congestion control named “HRF (Hybrid RTT-Fair)-TCP”. Today, it is a serious problem that the flows having smaller RTT utilize more bandwidth than others when multiple flows having different RTT values compete in the same network. This means that a user with longer RTT may not be able to obtain sufficient bandwidth by the current methods. This RTT fairness issue has been discussed in many TCP papers. An example is CR (Constant Rate) algorithm, which achieves RTT-fairness by multiplying the square of RTT value in its window increment phase against TCP-Reno. However, the method halves its windows size same as TCP-Reno when a packet loss is detected. This makes worse its efficiency in certain network cases. On the other hand, recent proposed TCP versions essentially require throughput efficiency and TCP-friendliness with TCP-Reno. Therefore, we try to keep these advantages in our TCP design in addition to RTT-fairness. In this paper, we make intuitive analytical models in which we separate resource utilization processes into two cases: utilization of bottleneck link capacity and that of buffer space at the bottleneck link router. These models take into account three characteristic algorithms (Reno, Constant Rate, Constant Increase) in window increment phase where a sender receives an acknowledgement successfully. Their validity is proved by both simulations and implementations. From these analyses, we propose HRF-TCP which switches two modes according to observed RTT values and achieves RTT fairness. Experiments are carried out to validate the proposed method. Finally, HRF-TCP outperforms conventional methods in RTT-fairness, efficiency and friendliness with TCP-Reno.

The demand of using applications that assume bidirectional communication such as voice telephony and peer-to-peer using wireless stations has been increasing and especially, the rapid increase of uplink traffic from wireless terminals is expected. However, in uplink WLANs, the hidden-station problem remains to be solved. In this paper, we point out this hidden-station problem and clarify the following unfairness between UDP and TCP uplink flows: 1) the effect of collision caused by hidden-station relationship on throughput and 2) the instability of the throughput depending on the number of hidden stations. To solve these problems, we propose a virtual multi-AP access mechanism. Our mechanism first groups stations according to the hidden-station relationship and type of transport protocol they use then assigns a virtually isolated channel to each group, which enables STAs to communicate as if STAs in different groups are connected to different isolated APs (virtual APs: VAPs). It can mitigate the effect caused by collisions between hidden stations and eliminate the contention between UDP and TCP uplink flows. Its performance is shown through simulation.

An important issue in designing a TCP congestion control algorithm is that it should allow the protocol to quickly adjust the end-to-end communication rate to the bandwidth on the bottleneck link. However, the TCP congestion control may function poorly in high bandwidth-delay product networks because of its slow response with large congestion windows. In this paper, we propose an enhanced version of TCP Vegas called Quick Vegas, in which we present an efficient congestion window control algorithm for a TCP source. Our algorithm improves the slow-start and congestion avoidance techniques of original Vegas. Simulation results show that Quick Vegas significantly improves the performance of connections as well as remaining fair when the bandwidth-delay product increases.

Many handover techniques in the Internet have been introduced with the development of mobile computing technologies. Although many proposed handover schemes utilize multiple wireless interfaces, having multiple wireless interfaces in a mobile device increases its power consumption, device installation space, and hardware costs. We have been studying handover schemes for mobile nodes with a single wireless interface. To achieve seamless and efficient handover, we focus on Stream Control Transmission Protocol (SCTP) that offers a message-oriented, reliable and connection-oriented delivery transport service. Unlike other transport protocols like TCP, SCTP can provide an end-to-end handover mechanism with multi-homing feature. However, the handover mechanism in the current SCTP causes large handover latency particularly when a mobile node has only one single wireless interface. This paper investigates the current issues of the SCTP handover mechanism, and proposes a new efficient handover scheme based on SCTP, which identifies a communication path as a pair of source and destination address. Additionally, we modified SCTP behavior when an SCTP endpoint received a SET PRIMARY message to change primary destination of peer endpoint. This paper shows that our scheme can reduce the handover latency by two to thirty seconds.

A transport layer mobility management scheme for handling seamless handoffs between appropriate networks is presented. The future mobile environment will be characterized by multimodal connectivity with dynamic switching. Many technologies have been proposed to support host mobility across diverse wireless networks, and operate in various layers of the network architecture. Our major focus is on the transport protocol that recovers packets lost during handoffs and controls transmission speed to achieve efficient communication. Majority of the existing technologies can maintain the connection by updating the information of a single connection around a handoff. Moreover, none of the studies extensively examine the handoff latencies and focus how an appropriate network is selected, during the handoff. In this paper, we first extensively investigate the various handoff latencies and discuss the limited performance of existing technologies based on the single connection. We then propose a new scheme resolving the problems by the transport protocol enabling the adaptive selection of an appropriate interface based on communication condition among all available interfaces. Finally, we demonstrate that the proposed scheme promptly and reliably selects the appropriate interface, and achieves excellent goodput performance by comparing with the existing technologies.

A critical design issue of Transmission Control Protocol (TCP) is its congestion control that allows the protocol to adjust the end-to-end communication rate based on the detection of packet loss. However, TCP congestion control may function poorly during its slow start and congestion avoidance phases. This is because TCP sends bursts of packets with the fast window increase and the ACK-clock based transmission in slow start, and respond slowly with large congestion windows especially in high bandwidth-delay product (BDP) networks during congestion avoidance. In this article, we propose an improved version of TCP, TCP-Ho, that uses an efficient congestion window control algorithm for a TCP source. According to the estimated available bandwidth and measured round-trip times (RTTs), the proposed algorithm adjusts the congestion window size with a rate between exponential growth and linear growth intelligently. Our extensive simulation results show that TCP-Ho significantly improves the performance of connections as well as remaining fair and stable when the BDP increases. Furthermore, it is feasible to implement because only sending part needs to be modified.

This paper considers the reliable multicast transport protocols used in hybrid networks that include wired and wireless networks and transparent proxy servers. We present four analytic performance models of two extreme reliable multicast transport protocols, sender-initiated and receiver-initiated, and supported and unsupported by transparent proxy servers are considered in each reliable multicast protocol. We analyze the throughputs of these four different models mathematically. Numerical results show that transparent proxy servers give good effects to overall performance. Furthermore, the receiver-initiated reliable multicast supported by transparent proxy servers gives better performances of total throughput than sender-initiated reliable multicast supported by transparent proxy servers. We provide efficiency criterion of the optimal number of transparent proxy servers for each protocol under varying wireless loss probabilities. Numerical results are verified by simulations.

It is well known that TCP does not fully utilize the available bandwidth in fast long-distance networks. To solve this scalability problem, several high speed transport protocols have been proposed. They include HighSpeed TCP (HS-TCP), Scalable TCP (S-TCP), Binary increase control TCP (BIC-TCP), and H-TCP. These protocols increase (decrease) their window size more aggressively (slowly) compared to standard TCP (STD-TCP). This paper aims at evaluating and comparing these high speed transport protocols through computer simulations. We select six metrics that are important for high speed protocols; scalability, buffer requirement, TCP friendliness, TCP compatibility, RTT fairness, and responsiveness. Simulation scenarios are carefully designed to investigate the performance of these protocols in terms of the metrics. Results clarify that each high speed protocol successfully solves the problem of STD-TCP. In terms of the buffer requirement, S-TCP and BIC-TCP have better performance. For TCP friendliness and compatibility, HS-TCP and H-TCP offer better performance. For RTT fairness, BIC-TCP and H-TCP are superior. For responsiveness, HS-TCP and H-TCP are preferred. However, H-TCP achieves a high degree of fairness at the expense of the link utilization. Thus, we understand that all the proposed high speed transport protocols have their own shortcomings. Thus, much more research is needed on high speed transport protocols.

Current TCP-friendly congestion control mechanisms adjust the packet rate in order to adapt to wired network conditions and obtain a throughput not exceeding that of a TCP connection operating under the same conditions. However, these mechanisms can not be directly applicable to wireless network because there is no way to distinguish congestion losses from wireless channel losses. In this letter, a new loss differentiation algorithm for wired-to-wireless streaming service is described. The approach does not only adjust the sending rate according to the network status, but also provide the useful feedback to the video encoder.

Current IP network has become the dominant paradigm for all networking environments. The significant cause of packet losses in such heterogenous networks is no longer limited to network congestion. Traditional TCP interprets every packet loss as caused by congestion which may be not the case in the current Internet. Misinterpretation of wireless random loss as an indication of network congestion results in TCP slowing down its sending rate unnecessarily. In this paper, we propose a new variant of TCP Vegas named RedVegas. By using the innate nature of Vegas and congestion indications marked by routers, RedVegas may detect random packet losses precisely. Through the packet loss differentiation, RedVegas reacts appropriately to the losses, and therefore the throughput of connection over heterogeneous networks can be significantly improved.