Equation-based Rate Control Research Articles

MAC-aware Loss Discrimination for Real-time Streaming Services in Wireless Networks

This paper addresses the rate control problem for real-time applications streamed over wireless networks. In wired networks, an equation-based rate control such as TCP-friendly rate control (TFRC) can be used to control the rates of a source under the assumption that the loss is primarily due to congestion. But in wireless networks, packet loss may be due either to congestion or to channel errors. Thus, it is necessary to differentiate between packet loss due to wireless channel errors and that due to congestion. The MAC-aware rate control scheme for real-time streaming applications discriminates packet losses due to channel errors using the event generation in the MAC layer. The simulation results show that the MAC-aware rate control scheme has higher throughput than the rate control scheme without loss classification.

A New Congestion Control Approach on TFRC Over Wired and Wireless Networks

For advanced streaming applications over wired-wireless networks TCP-Friendly Rate Control (TFRC) has been widely adopted nowadays to give smooth sending rate and unceasing quality in streaming applications TFRC applies an equation-based rate control scheme. However, TFRC tends to fail in wireless environment if packet lost event was done by poor channel quality but network congestion. Therefore, TFRC not able to provide high quality-of-service for streaming applications over wired-wireless networks. In this paper, we proposed a delay based uni-directional delay jitter based TFRC with end-to-end semantic over wired-wireless networks. This scheme provide smooth sending rate and TCP friendly characteristics like standard TFRC, even it also increase the throughput by estimating the available bandwidth in wired-wireless networks with bursty nature of background traffic. Simulation results show performance improvement without intrusiveness issue and even if background traffic is bursty over wired-wireless networks.

Equation Based Rate Control and Multiple Connections for Adaptive Video Streaming in Cellular Networks

Equation Based Rate Control and Multiple Connections for Adaptive Video Streaming in Cellular Networks

Extending equation-based congestion control to high-speed and long-distance networks

TCP-friendly rate control (TFRC), an equation-based congestion control protocol, has been a promising alternative to TCP for multimedia streaming applications. However, TFRC using the TCP response function, has the same poor performance as TCP in high-speed and long-distance networks. In this paper, we propose high-speed equation-based rate control (HERC), as an extension of TFRC by replacing the TCP response function with a high-speed response function. HERC could be used for applications, such as high-definition video streaming, and remote collaboration involving high-resolution visualization, which prefer a high-speed and relatively smooth sending rate. The impact of a general high-speed response function on the throughput and smoothness of HERC is studied analytically and verified by using simulation. Our result indicates that by using the response function of a high-speed TCP variant and tuning HERC parameters accordingly, HERC can compete fairly with high-speed TCP flows in the same network, while maintaining the desired smoothness of TFRC.

ARTP: A buffer-aware rate control protocol for media streaming

We present adaptive real-time transport protocol (ARTP), a media streaming transport protocol that implements a congestion control mechanism. With this mechanism, the sender adapts its sending rate to network conditions and to the buffering capacity of the receiver. This adaptiveness takes into account the real-time constraints of media streaming. It aims at ensuring media playback continuity, and at achieving a low packet loss rate during media streaming sessions. ARTP ensures the continuity of media playback by buffering media packets during congestion-free periods and reduces the loss rate by reducing the transmission rate during congestion periods. This protocol considers the size of the buffer that the receiver dedicates to rate control in order to avoid overflow or underflow of the buffer. This approach allows limited memory devices such as cellular phones and PDAs to take advantage of rate control. ARTP is based on the feedback that the real-time control protocol (RTCP) reports give with the addition of two new parameters that we define in this paper: the steady state loss event rate and the duration to the next feedback report. It also requires that the real-time streaming protocol (RTSP) provide the server with the size of the buffer that the client dedicates to rate control. Our NS-2 simulations show that, besides buffer protection, ARTP significantly reduces the loss rate. Compared to Additive Increase Multiplicative Decrease (AIMD) rate control techniques, ARTP provides a better media quality by ensuring the continuity of media playback and, compared to equation-based rate control techniques, it achieves a better loss rate and reduces the bandwidth used for feedback.

Multiple TFRC Connections Based Rate Control for Wireless Networks

Rate control is an important issue in video streaming applications for both wired and wireless networks. A widely accepted rate control method in wired networks is equation based rate control , in which the TCP friendly rate is determined as a function of packet loss rate, round trip time and packet size. This approach, also known as TCP friendly rate control (TFRC), assumes that packet loss in wired networks is primarily due to congestion, and as such is not applicable to wireless networks in which the bulk of packet loss is due to error at the physical layer. In this paper, we propose multiple TFRC connections as an end-to-end rate control solution for wireless video streaming. We show that this approach not only avoids modifications to the network infrastructure or network protocol, but also results in full utilization of the wireless channel. NS-2 simulations, actual experiments over 1 times RTT CDMA wireless data network, and and video streaming simulations using traces from the actual experiments, are carried out to validate, and characterize the performance of our proposed approach

Equation based rate control scheme for video streaming over wireless channels with link level ARQ

Equation based TCP-friendly rate control (TFRC) protocol has been proposed to support video streaming applications. In order to improve TFRC performance in wireless channels, the link level automatic repeat request (ARQ) scheme is usually deployed. However, ARQ cannot ensure strict delay guarantees, especially over multihop links. This paper introduces a theoretical model to deduce an equation for packet size adjustment in transport layer to minimize retransmission delay by taking into consideration the causative reasons inducing retransmission in link layer. An enhanced TFRC (ETFRC) scheme is proposed integrating TFRC with variable packet size policy. Simulation results demonstrate that higher goodput, lower packet loss rate (PLR), lower frame transmission delay and jitter with good fairness can be achieved by our proposed mechanism.

Rate control for streaming video over wireless

Rate control is an important issue in video streaming applications for both wired and wireless networks. A widely accepted rate control method in wired networks is TCP-friendly rate control (TFRC) (Floyd, 2000). It is equation-based rate control in which the TCP-friendly rate is determined as a function of packet loss rate, round-trip time, and packet size. TFRC assumes that packet loss in wired networks is primarily due to congestion, and as such is not applicable to wireless networks in which the main cause of packet loss is at the physical layer. In this article we review existing approaches to solve this problem. Then we propose multiple TFRC connections as an end-to-end rate control solution for wireless video streaming. We show that this approach not only avoids modifications to the network infrastructure or network protocol, but also results in full utilization of the wireless channel. NS-2 simulations, actual experiments over a 1/spl times/RTT CDMA wireless data network, and video streaming simulations using traces from the actual experiments are carried out to characterize the performance and show the efficiency of our proposed approach.

On the long-run behavior of equation-based rate control

We consider unicast equation-based rate control, where, at some points in time, a source adjusts its rate to f(p,r). Here p is an on-line estimate of the loss-event rate, r, of the mean round-trip time, both as observed by this source, and f is a TCP throughput formula. It was generally believed that such a source would be TCP-friendly, that is, under the same operating conditions, its long-run time-average send rate (throughput) would not be larger than that of a TCP source. Our goal is to identify whether, and how far, this is true. First, we identify factors that play a role in TCP friendliness and find that it is important to study them separately. Then we analyze the importance of individual factors. A first factor is conservativeness (= throughput not larger than f(p,r)). We show that conservativeness is influenced by some convexity properties of f(p,r) with respect to p, and the covariance of the loss process. We show that in many real life cases these conditions result in conservativeness and, sometimes, excessive conservativeness. This explains the previously observed phenomena of throughput-drop when losses are high and f is the so-called PFTK formula. The second factor is that the source may experience considerably different loss-event rate than a TCP source. We identify and analyze two limit cases where this may lead to either TCP-friendliness or, in contrast, non-TCP-friendliness. Other factors such as round trip time and obedience of TCP to its own formula are found to be less significant. Our claims are obtained by analysis, and verified by numerical examples, simulations, laboratory and Internet experiments. Our results suggest that TCP-friendliness is difficult to verify in practice, whereas conservativeness is easier.

Modeling the throughput of TCP Vegas

Previous analytic models of TCP Vegas throughput have been developed for loss-free (all-Vegas) networks. This work develops a simple and accurate analytic model for the throughput of a TCP Vegas bulk transfer in the presence of packet loss, as a function of average round trip time, minimum round trip time, and loss rate for the transfer. Similar models have previously been developed for TCP Reno. However, several aspects of TCP Vegas need to be treated differently than their counterparts in Reno. The proposed model captures the key innovative mechanisms that Vegas employs during slow start, congestion avoidance, and congestion recovery. The results include (1) a simple, validated model of TCP Vegas throughput that can be used for equation-based rate control of other flows such as UDP streams, (2) a simple formula to determine, from the measured packet loss rate, whether the network buffers are overcommitted and thus the TCP Vegas flow cannot reach the specified target lower threshold on throughput, (3) new insights into the design and performance of TCP Vegas, and (4) comparisons between TCP Vegas and TCP Reno including new insights regarding incremental deployment of TCP Vegas.

Wave and equation based rate control using multicast round trip time

This paper introduces Wave and Equation Based Rate Control (WEBRC), the first multiple rate multicast congestion control protocol to be equation based. The equation-based approach enforces fairness to TCP with the benefit that fluctuations in the flow rate are small in comparison to TCP.This paper also introduces the multicast round trip time (MRTT), a multicast analogue of the unicast round trip time (RTT). The MRTT is fundamental to the equation-based protocol that each receiver uses to adjust its reception rate. Each receiver independently measures its own MRTT without placing any added messaging burden on the receiver, the sender or the intermediate network elements. Benefits provided by the MRTT include those that the RTT provides to TCP, e.g., reduced reception rates in reaction to buffer filling and fair sharing of bottleneck links. In addition, the use of MRTT is shown to synchronize and equalize the reception rates of proximate receivers and to cause reception rates to increase as the density of receivers increases.Another innovation of WEBRC is the idea of transmitting data with waves: the transmission rate on a channel is periodic, with an exponentially decreasing form during an active period followed by a quiescent period. Benefits of using waves include insensitivity to large IGMP leave latency; a frequency of joins and leaves by each receiver that is small and independent of the receiver reception rate; the use of a small number of multicast channels; fine-grained control over the receiver reception rate; and minimal, at times nonexistent, losses due to buffer overflow.

On the long-run behavior of equation-based rate control

We consider unicast equation-based rate control, where a source estimates the loss event ratio $p$, and, primarily at loss events, adjusts its send rate to $f(p)$. Function $f$ is assumed to represent the loss-throughput relation that TCP would experience. When no loss occurs, the rate may also be increased according to some additional mechanism. We assume that the loss event interval estimator is non-biased. If the loss process is deterministic, the control is TCP-friendly in the long-run, i.e, the average throughput does not exceed that of TCP. If, in contrast, losses are random, it is a priori not clear whether this holds, due to the non-linearity of $f$, and a phenomenon similar to Feller's paradox. Our goal is to identify the key factors that drive whether, and how far, the control is TCP friendly (in the long run). As TCP and our source may experience different loss event intervals, we distinguish between TCP-friendliness and conservativeness (throughput does not exceed $f(p)$). We give a representation of the long term throughput, and derive that conservativeness is primarily influenced by various convexity properties of $f$, the variability of loss events, and the correlation structure of the loss process. In many cases, these factors lead to conservativeness, but we show reasonable experiments where the control is clearly non-conservative. However, our analysis also suggests that our source should experience a higher loss event ratio than TCP, which would make non-TCP friendliness less likely. Our findings provide guidelines that help understand when an equation base control is indeed TCP-friendly in the long-run, and in some cases, excessively so. The effects of round trip time and its variations are not included in this study.

Video multicast using layered FEC and scalable compression

The use of scalable video with layered multicast has been shown to be an effective method to achieve rate control in heterogeneous networks. We propose the use of layered forward error correction (FEC) as an error-control mechanism in a layered multicast framework. By organizing FEC into multiple layers, receivers can obtain different levels of protection commensurate with their respective channel conditions. Efficient network utilization is achieved as FEC streams are multicast, and only to receivers that need them. Furthermore, FEC is used without overall rate expansion by selectively dropping data layers to make room for FEC layers. Effects of bursty losses are amortized by staggering the FEC streams in time, giving rise to a tradeoff between delay and quality. For rate control at the receivers, we propose an equation-based approach that computes network usage as a function of measured network characteristics. We show that equation-based rate control achieves more fair bandwidth sharing amongst competing sessions as compared to existing multicast rate control schemes such as RLM and RLC. Fairness is achieved since competing sessions sharing a path will measure similar network characteristics. Simulations and actual MBONE experiments are performed using error-resilient, scalable video compression. We find that video quality is significantly improved at the same communication rate when layered FEC is used.