Abstract: Transmission Control Protocol (TCP) among the wired and wireless networks, it still has a practical problem; where the congestion control mechanism does not permit the data stream to get complete bandwidth over the existing network links. To solve this problem, many TCP protocols have been introduced with high speed performance. Therefore, an enhanced congestion window (cwnd) for the congestion control mechanism is proposed in this article to improve the performance of TCP by increasing the number of cycles of the new window to improve the transmitted packet number. The proposed algorithm used a new mechanism based on the available bandwidth of the connection to detect the capacity of network path in order to improve the regular clocking of congestion avoidance mechanism. The work in this paper based on using Network Simulator 2 (NS-2) to simulate the proposed algorithm.
Abstract: Various fairness models and criteria proposed by academia and industries for wired networks can be applied for ad hoc wireless network. The end-to-end fairness in an ad hoc wireless network is a challenging task compared to wired networks, which has not been addressed effectively. Most of the traffic in an ad hoc network are transport layer flows and thus the fairness of transport layer flows has attracted the interest of the researchers. The factors such as MAC protocol, routing protocol, the length of a route, buffer size, active queue management algorithm and the congestion control algorithms affects the fairness of transport layer flows. In this paper, we have considered the rate of data transmission, the queue management and packet scheduling technique. The ad hoc network is dynamic in nature due to various parameters such as transmission of control packets, multihop nature of forwarding packets, changes in source and destination nodes, changes in the routing path influences determining throughput and fairness among the concurrent flows. In addition, the effect of interaction between the protocol in the data link and transport layers has also plays a role in determining the rate of the data transmission. We maintain queue for each flow and the delay information of each flow is maintained accordingly. The pre-processing of flow is done up to the network layer only. The source and destination address information is used for separating the flow and the transport layer information is not used. This minimizes the delay in the network. Each flow is attached to a timer and is updated dynamically. Finite State Machine (FSM) is proposed for queue and transmission control mechanism. The performance of the proposed approach is evaluated in ns-2 simulation environment. The throughput and fairness based on mobility for different flows used as performance metrics. We have compared the performance of the proposed approach with ATP and the transport layer information is used. This minimizes the delay in the network. Each flow is attached to a timer and is updated dynamically. Finite State Machine (FSM) is proposed for queue and transmission control mechanism. The performance of the proposed approach is evaluated in ns-2 simulation environment. The throughput and fairness based on not mobility for different flows used as performance metrics. We have compared the performance of the proposed approach with ATP and MC-MLAS and the performance of the proposed approach is encouraging.
Abstract: In this paper, a robust decentralized congestion control strategy is developed for a large scale network with Differentiated Services (Diff-Serv) traffic. The network is modeled by a nonlinear fluid flow model corresponding to two classes of traffic, namely the premium traffic and the ordinary traffic. The proposed congestion controller does take into account the associated physical network resource limitations and is shown to be robust to the unknown and time-varying delays. Our proposed decentralized congestion control strategy is developed on the basis of Diff-Serv architecture by utilizing a robust adaptive technique. A Linear Matrix Inequality (LMI) condition is obtained to guarantee the ultimate boundedness of the closed-loop system. Numerical simulation implementations are presented by utilizing the QualNet and Matlab software tools to illustrate the effectiveness and capabilities of our proposed decentralized congestion control strategy.
Abstract: This work presents the Risk Threshold RED (RTRED)
congestion control strategy for TCP networks. In addition to the
maximum and minimum thresholds in existing RED-based strategies,
we add a third dropping level. This new dropping level is the risk
threshold which works with the actual and average queue sizes to
detect the immediate congestion in gateways. Congestion reaction
by RTRED is on time. The reaction to congestion is neither too
early, to avoid unfair packet losses, nor too late to avoid packet
dropping from time-outs. We compared our novel strategy with RED
and ARED strategies for TCP congestion handling using a NS-2
simulation script. We found that the RTRED strategy outperformed
RED and ARED.
Abstract: Reliable secure multicast communication in mobile
adhoc networks is challenging due to its inherent characteristics of
infrastructure-less architecture with lack of central authority, high
packet loss rates and limited resources such as bandwidth, time and
power. Many emerging commercial and military applications require
secure multicast communication in adhoc environments. Hence key
management is the fundamental challenge in achieving reliable
secure communication using multicast key distribution for mobile
adhoc networks. Thus in designing a reliable multicast key
distribution scheme, reliability and congestion control over
throughput are essential components. This paper proposes and
evaluates the performance of an enhanced optimized multicast cluster
tree algorithm with destination sequenced distance vector routing
protocol to provide reliable multicast key distribution. Simulation
results in NS2 accurately predict the performance of proposed
scheme in terms of key delivery ratio and packet loss rate under
varying network conditions. This proposed scheme achieves
reliability, while exhibiting low packet loss rate with high key
delivery ratio compared with the existing scheme.
Abstract: This paper presents design trade-off and performance impacts of
the amount of pipeline phase of control path signals in a wormhole-switched
network-on-chip (NoC). The numbers of the pipeline phase of the control
path vary between two- and one-cycle pipeline phase. The control paths
consist of the routing request paths for output selection and the arbitration
paths for input selection. Data communications between on-chip routers are
implemented synchronously and for quality of service, the inter-router data
transports are controlled by using a link-level congestion control to avoid
lose of data because of an overflow. The trade-off between the area (logic
cell area) and the performance (bandwidth gain) of two proposed NoC router
microarchitectures are presented in this paper. The performance evaluation is
made by using a traffic scenario with different number of workloads under
2D mesh NoC topology using a static routing algorithm. By using a 130-nm
CMOS standard-cell technology, our NoC routers can be clocked at 1 GHz,
resulting in a high speed network link and high router bandwidth capacity
of about 320 Gbit/s. Based on our experiments, the amount of control path
pipeline stages gives more significant impact on the NoC performance than
the impact on the logic area of the NoC router.
Abstract: This paper is aimed at describing a delay-based endto-
end (e2e) congestion control algorithm, called Very FAST TCP
(VFAST), which is an enhanced version of FAST TCP. The main
idea behind this enhancement is to smoothly estimate the Round-Trip
Time (RTT) based on a nonlinear filter, which eliminates throughput
and queue oscillation when RTT fluctuates. In this context, an evaluation
of the suggested scheme through simulation is introduced, by
comparing our VFAST prototype with FAST in terms of throughput,
queue behavior, fairness, stability, RTT and adaptivity to changes in
network. The achieved simulation results indicate that the suggested
protocol offer better performance than FAST TCP in terms of RTT
estimation and throughput.
Abstract: The use of buffer thresholds, blocking and adequate
service strategies are well-known techniques for computer networks
traffic congestion control. This motivates the study of series queues
with blocking, feedback (service under Head of Line (HoL) priority
discipline) and finite capacity buffers with thresholds. In this paper,
the external traffic is modelled using the Poisson process and the
service times have been modelled using the exponential distribution.
We consider a three-station network with two finite buffers, for
which a set of thresholds (tm1 and tm2) is defined. This computer
network behaves as follows. A task, which finishes its service at
station B, gets sent back to station A for re-processing with
probability o. When the number of tasks in the second buffer exceeds
a threshold tm2 and the number of task in the first buffer is less than
tm1, the fed back task is served under HoL priority discipline. In
opposite case, for fed backed tasks, “no two priority services in
succession" procedure (preventing a possible overflow in the first
buffer) is applied. Using an open Markovian queuing schema with
blocking, priority feedback service and thresholds, a closed form
cost-effective analytical solution is obtained. The model of servers
linked in series is very accurate. It is derived directly from a twodimensional
state graph and a set of steady-state equations, followed
by calculations of main measures of effectiveness. Consequently,
efficient expressions of the low computational cost are determined.
Based on numerical experiments and collected results we conclude
that the proposed model with blocking, feedback and thresholds can
provide accurate performance estimates of linked in series networks.
Abstract: In this paper we investigated a number of the Internet
congestion control algorithms that has been developed in the last few
years. It was obviously found that many of these algorithms were
designed to deal with the Internet traffic merely as a train of
consequent packets. Other few algorithms were specifically tailored
to handle the Internet congestion caused by running media traffic that
represents audiovisual content. This later set of algorithms is
considered to be aware of the nature of this media content. In this
context we briefly explained a number of congestion control
algorithms and hence categorized them into the two following
categories: i) Media congestion control algorithms. ii) Common
congestion control algorithms. We hereby recommend the usage of
the media congestion control algorithms for the reason of being
media content-aware rather than the other common type of
algorithms that blindly manipulates such traffic. We showed that the
spread of such media content-aware algorithms over Internet will
lead to better congestion control status in the coming years. This is
due to the observed emergence of the era of digital convergence
where the media traffic type will form the majority of the Internet
traffic.
Abstract: Dedicated Short Range Communication (DSRC) is a
key enabling technology for the next generation of
communication-based safety applications. One of the important
problems for DSRC deployment is maintaining high performance
under heavy channel load. Many studies focus on congestion control
mechanisms for simulating hundreds of physical radios deployed on
vehicles. The U.S. department of transportation-s (DOT) Intelligent
Transportation Systems (ITS) division has a plan to chosen prototype
on-board devices capable of transmitting basic “Here I am" safety
messages to other vehicles. The devices will be used in an IntelliDrive
safety pilot deployment of up to 3,000 vehicles. It is hard to log the
information of 3,000 vehicles. In this paper we present the designs and
issues related to the DSRC Radio Testbed under heavy channel load.
The details not only include the architecture of DSRC Radio Testbed,
but also describe how the Radio Interfere System is used to help for
emulating the congestion radio environment.
Abstract: Continuously growing needs for Internet applications
that transmit massive amount of data have led to the emergence of
high speed network. Data transfer must take place without any
congestion and hence feedback parameters must be transferred from
the receiver end to the sender end so as to restrict the sending rate in
order to avoid congestion. Even though TCP tries to avoid
congestion by restricting the sending rate and window size, it never
announces the sender about the capacity of the data to be sent and
also it reduces the window size by half at the time of congestion
therefore resulting in the decrease of throughput, low utilization of
the bandwidth and maximum delay. In this paper, XCP protocol is
used and feedback parameters are calculated based on arrival rate,
service rate, traffic rate and queue size and hence the receiver
informs the sender about the throughput, capacity of the data to be
sent and window size adjustment, resulting in no drastic decrease in
window size, better increase in sending rate because of which there is
a continuous flow of data without congestion. Therefore as a result of
this, there is a maximum increase in throughput, high utilization of
the bandwidth and minimum delay. The result of the proposed work
is presented as a graph based on throughput, delay and window size.
Thus in this paper, XCP protocol is well illustrated and the various
parameters are thoroughly analyzed and adequately presented.
Abstract: Wireless ad hoc nodes are freely and dynamically
self-organize in communicating with others. Each node can act as
host or router. However it actually depends on the capability of
nodes in terms of its current power level, signal strength, number
of hops, routing protocol, interference and others. In this research,
a study was conducted to observe the effect of hops count over
different network topologies that contribute to TCP Congestion
Control performance degradation. To achieve this objective, a
simulation using NS-2 with different topologies have been
evaluated. The comparative analysis has been discussed based on
standard observation metrics: throughput, delay and packet loss
ratio. As a result, there is a relationship between types of topology
and hops counts towards the performance of ad hoc network. In
future, the extension study will be carried out to investigate the
effect of different error rate and background traffic over same
topologies.
Abstract: As originally designed for wired networks, TCP (transmission control protocol) congestion control mechanism is triggered into action when packet loss is detected. This implicit assumption for packet loss mostly due to network congestion does not work well in Mobile Ad Hoc Network, where there is a comparatively high likelihood of packet loss due to channel errors and node mobility etc. Such non-congestion packet loss, when dealt with by congestion control mechanism, causes poor TCP performance in MANET. In this study, we continue to investigate the impact of the interaction between transport protocols and on-demand routing protocols on the performance and stability of 802.11 multihop networks. We evaluate the important wireless networking events caused routing change, and propose a cross layer method to delay the unnecessary routing changes, only need to add a sensitivity parameter α , which represents the on-demand routing-s reaction to link failure of MAC layer. Our proposal is applicable to the plain 802.11 networking environment, the simulation results that this method can remarkably improve the stability and performance of TCP without any modification on TCP and MAC protocol.
Abstract: Transmission control protocol (TCP) Vegas detects
network congestion in the early stage and successfully prevents
periodic packet loss that usually occurs in TCP Reno. It has been
demonstrated that TCP Vegas outperforms TCP Reno in many
aspects. However, TCP Vegas suffers several problems that affect its
congestion avoidance mechanism. One of the most important
weaknesses in TCP Vegas is that alpha and beta depend on a good
expected throughput estimate, which as we have seen, depends on a
good minimum RTT estimate. In order to make the system more
robust alpha and beta must be made responsive to network conditions
(they are currently chosen statically). This paper proposes a modified
Vegas algorithm, which can be adjusted to present good performance
compared to other transmission control protocols (TCPs). In order to
do this, we use PSO algorithm to tune alpha and beta. The simulation
results validate the advantages of the proposed algorithm in term of
performance.
Abstract: The importance of our country-s communication
system is noticeable when a disaster occurs. The communication
system in our country includes wired and wireless telephone
networks, radio, satellite system and more increasingly internet. Even
though our communication system is most extensive and dependable,
extreme conditions can put a strain on them. Interoperability between
heterogeneous wireless networks can be used to provide efficient
communication for emergency first response. IEEE 802.21 specifies
Media Independent Handover (MIH) services to enhance the mobile
user experience by optimizing handovers between heterogeneous
access networks. This paper presents an algorithm to improve
congestion control in MIH framework. It is analytically shown that
by including time factor in network selection we can optimize
congestion in the network.
Abstract: In this paper, we provide complete end-to-end delay analyses including the relay nodes for instant messages. Message Session Relay Protocol (MSRP) is used to provide congestion control for large messages in the Instant Messaging (IM) service. Large messages are broken into several chunks. These chunks may traverse through a maximum number of two relay nodes before reaching destination according to the IETF specification of the MSRP relay extensions. We discuss the current solutions of sending large instant messages and introduce a proposal to reduce message flows in the IM service. We consider virtual traffic parameter i.e., the relay nodes are stateless non-blocking for scalability purpose. This type of relay node is also assumed to have input rate at constant bit rate. We provide a new scheduling policy that schedules chunks according to their previous node?s delivery time stamp tags. Validation and analysis is shown for such scheduling policy. The performance analysis with the model introduced in this paper is simple and straight forward, which lead to reduced message flows in the IM service.
Abstract: Congestion control is one of the fundamental issues in computer networks. Without proper congestion control mechanisms there is the possibility of inefficient utilization of resources, ultimately leading to network collapse. Hence congestion control is an effort to adapt the performance of a network to changes in the traffic load without adversely affecting users perceived utilities. AIMD (Additive Increase Multiplicative Decrease) is the best algorithm among the set of liner algorithms because it reflects good efficiency as well as good fairness. Our control model is based on the assumption of the original AIMD algorithm; we show that both efficiency and fairness of AIMD can be improved. We call our approach is New AIMD. We present experimental results with TCP that match the expectation of our theoretical analysis.