2,625 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
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Effective video multicast over wireless internet
With the rapid growth of wireless networks and great success of Internet video, wireless video services are expected to be widely deployed in the near future. As different types of wireless networks are converging into all IP networks, i.e., the Internet, it is important to study video delivery over the wireless Internet. This paper proposes a novel end-system based adaptation protocol calledWireless Hybrid Adaptation Layered Multicast (WHALM) protocol for layered video multicast over wireless Internet. In WHALM the sender dynamically collects bandwidth distribution from the receivers and uses an optimal layer rate allocation mechanism to reduce the mismatches between the coarse-grained layer subscription levels and the heterogeneous and dynamic rate requirements from the receivers, thus maximizing the degree of satisfaction of all the receivers in a multicast session. Based on sampling theory and theory of probability, we reduce the required number of bandwidth feedbacks to a reasonable degree and use a scalable feedback mechanism to control the feedback process practically. WHALM is also tuned to perform well in wireless networks by integrating an end-to-end loss differentiation algorithm (LDA) to differentiate error losses from congestion losses at the receiver side. With a series of simulation experiments over NS platform, WHALM has been proved to be able to greatly improve the degree of satisfaction of all the receivers while avoiding congestion collapse on the wireless Internet
Q-AIMD: A Congestion Aware Video Quality Control Mechanism
Following the constant increase of the multimedia traffic, it seems necessary to allow transport protocols to be aware of the video quality of the transmitted flows rather than the throughput. This paper proposes a novel transport mechanism adapted to video flows. Our proposal, called Q-AIMD for video quality AIMD (Additive Increase Multiplicative Decrease), enables fairness in video quality while transmitting multiple video flows. Targeting video quality fairness allows improving the overall video quality for all transmitted flows, especially when the transmitted videos provide various types of content with different spatial resolutions. In addition, Q-AIMD mitigates the occurrence of network congestion events, and dissolves the congestion whenever it occurs by decreasing the video quality and hence the bitrate. Using different video quality metrics, Q-AIMD is evaluated with different video contents and spatial resolutions. Simulation results show that Q-AIMD allows an improved overall video quality among the multiple transmitted video flows compared to a throughput-based congestion control by decreasing significantly the quality discrepancy between them
GTFRC, a TCP friendly QoS-aware rate control for diffserv assured service
This study addresses the end-to-end congestion control support over the DiffServ Assured Forwarding (AF) class. The resulting Assured Service (AS) provides a minimum level of throughput guarantee. In this context, this article describes a new end-to-end mechanism for continuous transfer based on TCP-Friendly Rate Control (TFRC). The proposed approach modifies TFRC to take into account the QoS negotiated. This mechanism, named gTFRC, is able to reach the minimum throughput guarantee whatever the flow’s RTT and target rate. Simulation measurements and implementation over a real QoS testbed demonstrate the efficiency of this mechanism either in over-provisioned or exactly-provisioned network. In addition, we show that the gTFRC mechanism can be used in the same DiffServ/AF class with TCP or TFRC flows
Towards a sender-based TCP friendly rate control (TFRC) protocol
Pervasive communications are increasingly sent over mobile devices and personal digital assistants. This trend is currently observed by mobile phone service providers which have measured a significant increase in multimedia traffic. To better carry multimedia traffic, the IETF standardized a new TCP Friendly Rate Control (TFRC) protocol. However, the current receiver-based TFRC design is not well suited to resource limited end systems. In this paper, we propose a scheme to shift resource allocation and computation to the sender. This sender-based approach led us to develop a new algorithm for loss notification and loss-rate computation. We detail the complete implementation of a user-level prototype and demonstrate the gain obtained in terms of memory requirements and CPU processing compared to the current design. We also evaluate the performance obtained in terms of throughput smoothness and fairness with TCP and we note this shifting solves security issues raised by classical TFRC implementations
Design and analysis for TCP-friendly window-based congestion control
The current congestion control mechanisms for the Internet date back to the early 1980’s and were
primarily designed to stop congestion collapse with the typical traffic of that era. In recent years the
amount of traffic generated by real-time multimedia applications has substantially increased, and the
existing congestion control often does not opt to those types of applications. By this reason, the Internet
can be fall into a uncontrolled system such that the overall throughput oscillates too much by a single
flow which in turn can lead a poor application performance. Apart from the network level concerns,
those types of applications greatly care of end-to-end delay and smoother throughput in which the
conventional congestion control schemes do not suit. In this research, we will investigate improving the
state of congestion control for real-time and interactive multimedia applications. The focus of this work
is to provide fairness among applications using different types of congestion control mechanisms to get
a better link utilization, and to achieve smoother and predictable throughput with suitable end-to-end
packet delay
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