Improvements in DCCP congestion control for satellite links

We propose modifications in the TCP-Friendly Rate Control (TFRC) congestion control mechanism from the Datagram Congestion Control Protocol (DCCP) intended for use with real-time traffic, which are aimed at improving its performance for long delay (primarily satellite) links. Firstly, we propose an algorithm to optimise the number of feedback messages per round trip time (RTT) rather than use the currently standard of at least one per RTT, based on the observed link delay. We analyse the improvements achievable with proposed modification in different phases of congestion control and present results from simulations with modified ns-2 DCCP and live experiments using the modified DCCP Linux kernel implementation. We demonstrate that the changes results in improved slow start performance and a reduced data loss compared to standard DCCP, while the introduced overhead remains acceptable

Promoting the use of reliable rate based transport protocols: the Chameleon protocol

Rate-based congestion control, such as TFRC, has not been designed to enable reliability. Indeed, the birth of TFRC protocol has resulted from the need for a congestion-controlled transport protocol in order to carry multimedia traffic. However, certain applications still prefer the use of UDP in order to implement their own congestion control on top of it. The present contribution proposes to design and validate a reliable rate-based protocol based on the combined use of TFRC, SACK and an adapted flow control. We argue that rate-based congestion control is a perfect alternative to window-based congestion control as most of today applications need to interact with the transport layer and should not be only limited to unreliable services. In this paper, we detail the implementation of a reliable rate-based protocol named Chameleon and bring out to the networking community an ns-2 implementation for evaluation purpose

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

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

A novel multimedia adaptation architecture and congestion control mechanism designed for real-time interactive applications

PhDThe increasing use of interactive multimedia applications over the Internet has created a problem of congestion. This is because a majority of these applications do not respond to congestion indicators. This leads to resource starvation for responsive flows, and ultimately excessive delay and losses for all flows therefore loss of quality. This results in unfair sharing of network resources and increasing the risk of network ‘congestion collapse’. Current Congestion Control Mechanisms such as ‘TCP-Friendly Rate Control’ (TFRC) have been able to achieve ‘fair-share’ of network resource when competing with responsive flows such as TCP, but TFRC’s method of congestion response (i.e. to reduce Packet Rate) is not ideally matched for interactive multimedia applications which maintain a fixed Frame Rate. This mismatch of the two rates (Packet Rate and Frame Rate) leads to buffering of frames at the Sender Buffer resulting in delay and loss, and an unacceptable reduction of quality or complete loss of service for the end-user. To address this issue, this thesis proposes a novel Congestion Control Mechanism which is referred to as ‘TCP-friendly rate control – Fine Grain Scalable’ (TFGS) for interactive multimedia applications. This new approach allows multimedia frames (data) to be sent as soon as they are generated, so that the multimedia frames can reach the destination as quickly as possible, in order to provide an isochronous interactive service. This is done by maintaining the Packet Rate of the Congestion Control Mechanism (CCM) at a level equivalent to the Frame Rate of the Multimedia Encoder.The response to congestion is to truncate the Packet Size, hence reducing the overall bitrate of the multimedia stream. This functionality of the Congestion Control Mechanism is referred to as Packet Size Truncation (PST), and takes advantage of adaptive multimedia encoding, such as Fine Grain Scalable (FGS), where the multimedia frame is encoded in order of significance, Most to Least Significant Bits. The Multimedia Adaptation Manager (MAM) truncates the multimedia frame to the size indicated by the Packet Size Truncation function of the CCM, accurately mapping user demand to available network resource. Additionally Fine Grain Scalable encoding can offer scalability at byte level granularity, providing a true match to available network resources. This approach has the benefits of achieving a ‘fair-share’ of network resource when competing with responsive flows (as similar to TFRC CCM), but it also provides an isochronous service which is of crucial benefit to real-time interactive services. Furthermore, results illustrate that an increased number of interactive multimedia flows (such as voice) can be carried over congested networks whilst maintaining a quality level equivalent to that of a standard landline telephone. This is because the loss and delay arising from the buffering of frames at the Sender Buffer is completely removed. Packets sent maintain a fixed inter-packet-gap-spacing (IPGS). This results in a majority of packets arriving at the receiving end at tight time intervals. Hence, this avoids the need of using large Playout (de-jitter) Buffer sizes and adaptive Playout Buffer configurations. As a result this reduces delay, improves interactivity and Quality of Experience (QoE) of the multimedia application

The performance of DCCP TCP-like with initial slow-start threshold manipulation

This paper investigates the performance of the implementation of modified initial slow-start threshold size in Datagram Congestion Control Protocol (DCCP) TCP-like (CCID-2) over long delay link network.TCP-like is one of a congestion control mechanism for DCCP which is suitable for the delivery of multimedia data with abrupt changes during the transmission. The scenario is set for long delay link network,where the impact of the modified slow-start threshold value in TCP-like is significant.As a result, we managed to reduce the time required to obtain the maximum throughput in TCP-like during in the slow-start phase.The result shows that with the correct manipulation of initial slow-start threshold size for TCP-like, it will give a significant improvement to TCP-like performance over long delay link where the maximum throughput during the slow-start phase can be achieved faster

TCP-Friendly Rate Control with Token Bucket for VoIP Congestion Control

TCP Friendly Rate Control (TFRC) is a congestion control algorithm that provides a smooth transmission rate for real-time network applications. TFRC refrains from halving the sending rate on every packet drop, instead it is adjusted as a function of the loss rate during a single round trip time. TFRC has been proven to be fair when competing with TCP flows over congested links, but it lacks quality-of-service parameters to improve the performance of real-time traffic. A problem with TFRC is that it uses additive increase to adjust the sending rate during periods with no congestion. This leads to short term congestion that can degrade the quality of voice applications. We propose two changes to TFRC that improve the performance of VoIP applications. Our implementation, TFRC with Token Bucket (TFRC-TB), uses discrete calculated bit rates based on audio codec bandwidth usage to increase the sending rate. Also, it uses a token bucket to control the sending rate during congestion periods. We have used ns2, the network simulator, to compare our implementation to TFRC in a wide range of network conditions. Our results suggest that TFRC-TB can provide a quality of service (QoS) mechanism to voice applications while competing fairly with other traffic over congested links