Internal Model Hop-by-hop Congestion Control for High-Speed Networks

This paper presents a hop-by-hop congestion control for highspeed networks. The control policy relies on the data exchange between adjacent nodes of the network (nearest-neighbour interaction). The novelty of this paper consists in the extensive use of Internal Model Control (IMC) to set the rates of the traffic flows. As a result, the proposed congestion control provides upper-bounds of the queue lengths in all the network buffers (overflow avoidance), avoids wasting the assigned capacity (full link utilisation) and guarantees the congestion recovery. Numerical simulations prove the effectiveness of the scheme

On-Line End-to-End Congestion Control

Congestion control in the current Internet is accomplished mainly by TCP/IP. To understand the macroscopic network behavior that results from TCP/IP and similar end-to-end protocols, one main analytic technique is to show that the the protocol maximizes some global objective function of the network traffic. Here we analyze a particular end-to-end, MIMD (multiplicative-increase, multiplicative-decrease) protocol. We show that if all users of the network use the protocol, and all connections last for at least logarithmically many rounds, then the total weighted throughput (value of all packets received) is near the maximum possible. Our analysis includes round-trip-times, and (in contrast to most previous analyses) gives explicit convergence rates, allows connections to start and stop, and allows capacities to change.Comment: Proceedings IEEE Symp. Foundations of Computer Science, 200

SSthreshless Start: A Sender-Side TCP Intelligence for Long Fat Network

Measurement shows that 85% of TCP flows in the internet are short-lived flows that stay most of their operation in the TCP startup phase. However, many previous studies indicate that the traditional TCP Slow Start algorithm does not perform well, especially in long fat networks. Two obvious problems are known to impact the Slow Start performance, which are the blind initial setting of the Slow Start threshold and the aggressive increase of the probing rate during the startup phase regardless of the buffer sizes along the path. Current efforts focusing on tuning the Slow Start threshold and/or probing rate during the startup phase have not been considered very effective, which has prompted an investigation with a different approach. In this paper, we present a novel TCP startup method, called threshold-less slow start or SSthreshless Start, which does not need the Slow Start threshold to operate. Instead, SSthreshless Start uses the backlog status at bottleneck buffer to adaptively adjust probing rate which allows better seizing of the available bandwidth. Comparing to the traditional and other major modified startup methods, our simulation results show that SSthreshless Start achieves significant performance improvement during the startup phase. Moreover, SSthreshless Start scales well with a wide range of buffer size, propagation delay and network bandwidth. Besides, it shows excellent friendliness when operating simultaneously with the currently popular TCP NewReno connections.Comment: 25 pages, 10 figures, 7 table

OSCAR: A Collaborative Bandwidth Aggregation System

The exponential increase in mobile data demand, coupled with growing user expectation to be connected in all places at all times, have introduced novel challenges for researchers to address. Fortunately, the wide spread deployment of various network technologies and the increased adoption of multi-interface enabled devices have enabled researchers to develop solutions for those challenges. Such solutions aim to exploit available interfaces on such devices in both solitary and collaborative forms. These solutions, however, have faced a steep deployment barrier. In this paper, we present OSCAR, a multi-objective, incentive-based, collaborative, and deployable bandwidth aggregation system. We present the OSCAR architecture that does not introduce any intermediate hardware nor require changes to current applications or legacy servers. The OSCAR architecture is designed to automatically estimate the system's context, dynamically schedule various connections and/or packets to different interfaces, be backwards compatible with the current Internet architecture, and provide the user with incentives for collaboration. We also formulate the OSCAR scheduler as a multi-objective, multi-modal scheduler that maximizes system throughput while minimizing energy consumption or financial cost. We evaluate OSCAR via implementation on Linux, as well as via simulation, and compare our results to the current optimal achievable throughput, cost, and energy consumption. Our evaluation shows that, in the throughput maximization mode, we provide up to 150% enhancement in throughput compared to current operating systems, without any changes to legacy servers. Moreover, this performance gain further increases with the availability of connection resume-supporting, or OSCAR-enabled servers, reaching the maximum achievable upper-bound throughput

Rare: Resource Aware Routing for mEsh

An important element of any routing protocol used for Wireless Mesh Networks (WMNs) is the link cost function used to represent the radio link characteristic. The majority of the routing protocols for WMNs attempt to accurately characterise the radio link quality by constructing the link cost function from the measurements obtained using active probing techniques, which introduces overhead. In this paper we propose a new approach called Resource Aware Routing for mEsh (RARE) which instead employs passive monitoring to gather radio link information. This results in a smaller overhead than the other methods that require active network probing, and is load independent since it does not require an access to the medium. Moreover, se show that our RARE approach performs well in a real radio environment through a number of experiments performed on a static 17 node WLAN mesh testbed

NEIGHBOURHOOD LOAD ROUTING AND MULTI-CHANNELS IN WIRELESS MESH NETWORKS

As an emerging technology, wireless mesh networks are making significant progress in the area of wireless networks in recent years. Routing in Wireless Mesh Network (WMN) is challenging because of the unpredictable variations of the wireless environment. Traditional mechanisms have been proved that the routing performance would get deteriorated and ideal metrics must be explored. Most wireless routing protocols that are currently available are designed to use a single channel. The available network capacity can be increased by using multiple channels, but this requires the development of new protocols specifically designed for multi-channel operation. In this paper, we propose Neighbourhood load routing metric in single channel mesh networks and also present the technique to utilize multiple channels and multiple interfaces between routers for communication. The traditional routing metrics Hop Count and Weighted Cumulative Expected Transmission Time (WCETT) are used in routing. We compare performance of AODV-HOP, WCETT and NLR routing metrics in singlechannel and multichannel environment by considering throughput and end to end delay performance metrics. Our results show that NLR performs better in singlechannel environment

An Unobtrusive Method for Tracking Network Latency in Online Games

Online games are a very important class of distributed interactive applications. Their success is heavily dependant on the level of consistency that can be maintained between participants communicating in the virtual world. Achieving a high level of consistency usually involves the transmission of a large amount of network traffic. However, if the underlying network connecting participants is unable to process this traffic, then network latency will increase, which will in turn negatively impact on consistency. Many schemes exist which attempt to reduce network traffic, and thus reduce the effect of network latency on the interactive application. However, applications that employ these schemes tend to do so with little knowledge of the underlying network conditions, and assume a worst-case scenario of limited bandwidth. Such an assumption can actually cause these latency reduction schemes to perform sub-optimally, and ironically introduce more inconsistency than they reduce. Hence, it is important that online game applications become aware of network conditions, such as available bandwidth. Existing methods of estimating bandwidth operate by analysing trends in one-way latency, and require that extra data be transmitted between nodes in order to capture the latency trends. Such an approach does not suit online games, as the extra data requirements could increase network latency, and affect the ability of the application to scale to multiple participants. To deal with this issue, this paper proposes a method by which online games can unobtrusively track one-way network latency. This method requires no time-stamping information to be transmitted between participants and operates using data already being transmitted as part of the online game application, meaning that its impact on the network is minimal. NS2 simulations demonstrate that the trends collected by this method can be used to estimate bandwidth under certain conditions