Rate Control State-of-the-art Survey

The majority of Internet traffic use Transmission Control Protocol (TCP) as the transport level protocol. It provides a reliable ordered byte stream for the applications. However, applications such as live video streaming place an emphasis on timeliness over reliability. Also a smooth sending rate can be desirable over sharp changes in the sending rate. For these applications TCP is not necessarily suitable. Rate control attempts to address the demands of these applications. An important design feature in all rate control mechanisms is TCP friendliness. We should not negatively impact TCP performance since it is still the dominant protocol. Rate Control mechanisms are classified into two different mechanisms: window-based mechanisms and rate-based mechanisms. Window-based mechanisms increase their sending rate after a successful transfer of a window of packets similar to TCP. They typically decrease their sending rate sharply after a packet loss. Rate-based solutions control their sending rate in some other way. A large subset of rate-based solutions are called equation-based solutions. Equation-based solutions have a control equation which provides an allowed sending rate. Typically these rate-based solutions react slower to both packet losses and increases in available bandwidth making their sending rate smoother than that of window-based solutions. This report contains a survey of rate control mechanisms and a discussion of their relative strengths and weaknesses. A section is dedicated to a discussion on the enhancements in wireless environments. Another topic in the report is bandwidth estimation. Bandwidth estimation is divided into capacity estimation and available bandwidth estimation. We describe techniques that enable the calculation of a fair sending rate that can be used to create novel rate control mechanisms.Peer reviewe

Congestion Control for Streaming Media

The Internet has assumed the role of the underlying communication network for applications such as file transfer, electronic mail, Web browsing and multimedia streaming. Multimedia streaming, in particular, is growing with the growth in power and connectivity of today\u27s computers. These Internet applications have a variety of network service requirements and traffic characteristics, which presents new challenges to the single best-effort service of today\u27s Internet. TCP, the de facto Internet transport protocol, has been successful in satisfying the needs of traditional Internet applications, but fails to satisfy the increasingly popular delay sensitive multimedia applications. Streaming applications often use UDP without a proper congestion avoidance mechanisms, threatening the well-being of the Internet. This dissertation presents an IP router traffic management mechanism, referred to as Crimson, that can be seamlessly deployed in the current Internet to protect well-behaving traffic from misbehaving traffic and support Quality of Service (QoS) requirements of delay sensitive multimedia applications as well as traditional Internet applications. In addition, as a means to enhance Internet support for multimedia streaming, this dissertation report presents design and evaluation of a TCP-Friendly and streaming-friendly transport protocol called the Multimedia Transport Protocol (MTP). Through a simulation study this report shows the Crimson network efficiently handles network congestion and minimizes queuing delay while providing affordable fairness protection from misbehaving flows over a wide range of traffic conditions. In addition, our results show that MTP offers streaming performance comparable to that provided by UDP, while doing so under a TCP-Friendly rate

Rate adaptation for wireless video streaming based on error statistics

This paper presents a new rate-control algorithm for live video streaming over wireless IP networks, which is based on selective frame discarding. In the proposed mechanism excess 'P' frames are dropped from the output queue at the sender using a congestion estimate based on packet loss statistics obtained from RTCP feedback and from the Data Link (DL) layer. The performance of the algorithm is evaluated through computer simulation. This paper also presents a characterisation of packet losses owing to transmission errors and congestion, which can help in choosing appropriate strategies to maximise the video quality experienced by the end user. Copyright © 2007 Inderscience Enterprises Ltd

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

Adaptive Bitrate Streaming in Cloud Gaming

Cloud gaming streams games as video from a server to a client device making it susceptible to network congestion. Adaptive bitrate streaming estimates network capacity and sets encoding parameters to avoid exceeding the bandwidth of the connection. BBR is a congestion control algorithm as an alternative to current loss-based congestion control. We designed and implemented a bitrate adaptation heuristic based on BBR into GamingAnywhere, an open source cloud gaming platform. We conducted a user study and did objective analysis comparing our modified version to the original. Through our results, we found that our adaptive system was less challenging for players and improved retention rates and that there was no statistically significant difference in visual quality from objective testing

Congestion Control for Network-Aware Telehaptic Communication

Telehaptic applications involve delay-sensitive multimedia communication between remote locations with distinct Quality of Service (QoS) requirements for different media components. These QoS constraints pose a variety of challenges, especially when the communication occurs over a shared network, with unknown and time-varying cross-traffic. In this work, we propose a transport layer congestion control protocol for telehaptic applications operating over shared networks, termed as dynamic packetization module (DPM). DPM is a lossless, network-aware protocol which tunes the telehaptic packetization rate based on the level of congestion in the network. To monitor the network congestion, we devise a novel network feedback module, which communicates the end-to-end delays encountered by the telehaptic packets to the respective transmitters with negligible overhead. Via extensive simulations, we show that DPM meets the QoS requirements of telehaptic applications over a wide range of network cross-traffic conditions. We also report qualitative results of a real-time telepottery experiment with several human subjects, which reveal that DPM preserves the quality of telehaptic activity even under heavily congested network scenarios. Finally, we compare the performance of DPM with several previously proposed telehaptic communication protocols and demonstrate that DPM outperforms these protocols.Comment: 25 pages, 19 figure