2,257 research outputs found
Congestion Control using FEC for Conversational Multimedia Communication
In this paper, we propose a new rate control algorithm for conversational
multimedia flows. In our approach, along with Real-time Transport Protocol
(RTP) media packets, we propose sending redundant packets to probe for
available bandwidth. These redundant packets are Forward Error Correction (FEC)
encoded RTP packets. A straightforward interpretation is that if no losses
occur, the sender can increase the sending rate to include the FEC bit rate,
and in the case of losses due to congestion the redundant packets help in
recovering the lost packets. We also show that by varying the FEC bit rate, the
sender is able to conservatively or aggressively probe for available bandwidth.
We evaluate our FEC-based Rate Adaptation (FBRA) algorithm in a network
simulator and in the real-world and compare it to other congestion control
algorithms
Rate-Based End-to-End Congestion Control of Multimedia Traffic in Packet Switched Networks
This paper proposes an explicit rate-based end-to-end congestion control mechanism to alleviate congestion of multimedia traffic in packet switched networks such as the Internet. The congestion is controlled by adjusting the transmission rates of the sources in response to the feedback information from destination such as the buffer occupancy, packet arrival rate and service rate at the outgoing link, so that a desired quality of service (QoS) can be met. The QoS is defined in terms of packet loss ratio, transmission delay, power, and network utilization. Comparison studies demonstrate the effectiveness of the proposed scheme over New-Reno TCP (a variant of AIMD: additive increase multiplicative decrease) technique during simulated congestion. Since it is end-to-end, no router support is necessary, the proposed methodology can be readily applied to today\u27s Internet, as well as for real-time video and voice data transfer in unicast networks
Methods of Congestion Control for Adaptive Continuous Media
Since the first exchange of data between machines in different locations in early 1960s,
computer networks have grown exponentially with millions of people now using the
Internet. With this, there has also been a rapid increase in different kinds of services offered
over the World Wide Web from simple e-mails to streaming video. It is generally accepted
that the commonly used protocol suite TCP/IP alone is not adequate for a number of
modern applications with high bandwidth and minimal delay requirements. Many
technologies are emerging such as IPv6, Diffserv, Intserv etc, which aim to replace the onesize-fits-all approach of the current lPv4. There is a consensus that the networks will have
to be capable of multi-service and will have to isolate different classes of traffic through
bandwidth partitioning such that, for example, low priority best-effort traffic does not cause
delay for high priority video traffic. However, this research identifies that even within a
class there may be delays or losses due to congestion and the problem will require different
solutions in different classes.
The focus of this research is on the requirements of the adaptive continuous media
class. These are traffic flows that require a good Quality of Service but are also able to
adapt to the network conditions by accepting some degradation in quality. It is potentially
the most flexible traffic class and therefore, one of the most useful types for an increasing
number of applications.
This thesis discusses the QoS requirements of adaptive continuous media and
identifies an ideal feedback based control system that would be suitable for this class. A
number of current methods of congestion control have been investigated and two methods
that have been shown to be successful with data traffic have been evaluated to ascertain if
they could be adapted for adaptive continuous media. A novel method of control based on
percentile monitoring of the queue occupancy is then proposed and developed. Simulation
results demonstrate that the percentile monitoring based method is more appropriate to this
type of flow. The problem of congestion control at aggregating nodes of the network
hierarchy, where thousands of adaptive flows may be aggregated to a single flow, is then
considered. A unique method of pricing mean and variance is developed such that each
individual flow is charged fairly for its contribution to the congestion
Evaluation Study for Delay and Link Utilization with the New-Additive Increase Multiplicative Decrease Congestion Avoidance and Control Algorithm
As the Internet becomes increasingly heterogeneous, the issue of congestion
avoidance and control becomes ever more important. And the queue length,
end-to-end delays and link utilization is some of the important things in term
of congestion avoidance and control mechanisms. In this work we continue to
study the performances of the New-AIMD (Additive Increase Multiplicative
Decrease) mechanism as one of the core protocols for TCP congestion avoidance
and control algorithm, we want to evaluate the effect of using the AIMD
algorithm after developing it to find a new approach, as we called it the
New-AIMD algorithm to measure the Queue length, delay and bottleneck link
utilization, and use the NCTUns simulator to get the results after make the
modification for the mechanism. And we will use the Droptail mechanism as the
active queue management mechanism (AQM) in the bottleneck router. After
implementation of our new approach with different number of flows, we expect
the delay will less when we measure the delay dependent on the throughput for
all the system, and also we expect to get end-to-end delay less. And we will
measure the second type of delay a (queuing delay), as we shown in the figure 1
bellow. Also we will measure the bottleneck link utilization, and we expect to
get high utilization for bottleneck link with using this mechanism, and avoid
the collisions in the link
Energy-efficient wireless communication
In this chapter we present an energy-efficient highly adaptive network interface architecture and a novel data link layer protocol for wireless networks that provides Quality of Service (QoS) support for diverse traffic types. Due to the dynamic nature of wireless networks, adaptations in bandwidth scheduling and error control are necessary to achieve energy efficiency and an acceptable quality of service. In our approach we apply adaptability through all layers of the protocol stack, and provide feedback to the applications. In this way the applications can adapt the data streams, and the network protocols can adapt the communication parameters
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