This paper addresses the problem of streaming packetized media

As networked multimedia applications become widespread, it becomes increasingly important to ensure that these applications can coexist with current TCP-based applications. The TCP protocol is designed to reduce its sending rate when congestion is detected. Networked multimedia applications should exhibit similar behavior, if they wish to co-exist with TCP-based applications [9]. Using TCP for multimedia applications is not practical, since the protocol combines error control and congestion control, an appropriate combination for non-real time reliable data transfer, but inappropriate for loss-tolerant real time applications. In this paper we present a protocol that operates by measuring loss rates and round trip times and then uses them to set the transmission rate to that which TCP would achieve under similar conditions. The analysis in [13] is used to determine this "TCP-friendly" rate. This protocol represents a rst step towards developing a comprehensive protocol for congestion control for time-sensitive multimedia data streams. We evaluate the protocol under various tra c conditions, using simulations and implementation. The simulations are used to study the behavior of the protocol under controlled conditions. The implementation and experimentation involve over 300 experiments over the Internet, using several machines in the US and UK. Our experimental and simulation results show that the protocol is fair to TCP and to other sessions running TFRCP, and that the formula-based approach to achieving TCP-friendliness is indeed practical.

The IP multicast delivery mechanism provides a popular basis for delivery of continuous media to many participants in a conferencing application. However, the best-effort nature of multicast delivery results in poor playback quality in the presence of network congestion and packet loss. Contrary to widespread belief that the real-time nature of continuous media applications precludes the possibility of recovery of lost packets using retransmissions, we have found that these applications offer an interesting tradeoff between the desired playback quality and the desired degree of interactivity. In particular, we propose a new model of multicast delivery called resilient multicast in which each receiver in a multicast group can decide its own tradeoff between reliability and real-time requirements. To be effective, error recovery mechanisms in such a model need to be both fast (due to the real-time constraint) and have a low overhead (due to high volume of continuous media data). We have...

A new retransmission-based error control technique is presented that does not incur any additional latency in frame playout times, and hence are suitable for interactive applications. It takes advantage of the motion prediction loop employed in most motion compensationbased codecs. By correcting errors in a reference frame caused by earlier packet loss, it prevents error propagation. The technique rearranges the temporal dependency of frames so that a displayed frame is referenced for the decoding of its succeeding dependent frames much later than its display time. Thus, the delay in repairing lost packets can be effectively masked out. The developed technique is combined with layered video coding to maintain consistently good video quality even under heavy packet loss. Through the results of extensive Internet experiments, the paper shows that layered coding can be very effective when combined with the retransmissionbased error control technique for low-bit rate transmission over best...

While there is an increasing demand for streaming video applications on the Internet, various network characteristics make the deployment of these applications more challenging than traditional TCP-based applications like email and the Web. Packet loss can be detrimental to compressed video with interdependent frames because errors potentially propagate across many frames. While latency requirements do not permit retransmission of all lost data, we leverage the characteristics of MPEG-4 to selectively retransmit only the most important data in the bitstream. When latency constraints do not permit retransmission, we propose a mechanism for recovering this data using postprocessing techniques at the receiver. We quantify the effects of packet loss on the quality of MPEG-4 video, develop an analytical model to explain these effects, present a system to adaptively deliver MPEG-4 video in the face of packet loss and variable Internet conditions, and evaluate the effectiveness of the system under various network conditions.

A growing and important class of traffic in the Internet is so-called “streaming media,” in which a server transmits a packetized multimedia signal to a receiver that buffers the packets for playback. This playback buffer, if adequately sized, counteracts the adverse impact of delay jitter and reordering suffered by packets as they traverse the network, and if large enough also allows lost packets to be retransmitted before their playback deadline expires. We call this framework for retransmitting lost streaming-media packets “soft ARQ” since it represents a relaxed form of Automatic Repeat reQuest (ARQ). While state-of-the-art media servers employ such strategies, no work to date has proposed an optimal strategy for delay-constrained retransmissions of streaming media—specifically, one which determines what is the optimal packet to transmit at any given point in time. In this paper, we address this issue and present a framework for streaming media retransmission based on layered media representations, in which a signal is decomposed into a discrete number of layers and each successive layer provides enhanced quality. In our approach, the source chooses between transmitting (1) newer but critical coarse information (e.g., a first approximation of the media signal) and (2) older but less important refinement information (e.g., added details) using a decision process that minimizes the expected signal distortion at the receiver. To arrive at the proper mix of these two extreme strategies, we derive an optimal strategy for transmitting layered data over a binary erasure channel with instantaneous feedback. To provide a quantitative performance comparison of different transmission policies, we conduct a Markov-chain analysis, which shows that the best transmission policy is time-invariant and thus does not change as the frames’ layers approach their expiration times.

Scalable, or layered, media representation appears to be more suitable for transmission over the current heterogeneous networks. In this paper we study the problem of scalable layered streaming media delivery over a lossy channel. The goal is to find an optimal transmission policy to achieve the best playback quality at the client end. The problem involves some trade-offs such as time-constrained delivery and data dependencies. For example, a layer should be dropped before transmission if it already has a delay such that it cannot be played before its scheduled time. Moreover, less important layers with near-playback-time may also be dropped or delayed for delivery in order to save bandwidth for other layers with a high priority. We propose a framework for scalable streaming media delivery, that involves a novel scheduling algorithm called Expected runtime Distortion Based Scheduling, EDBS, which decides the order in which packets should be transmitted in order to improve client playback quality in the presence of channel losses. A fast greedy search algorithm is presented that achieves almost the same performance as an exhaustive search technique (98% of the time it results in the same schedule) with very low complexity and is applicable for real-time application.

Lack of QoS support in the Internet has not prevented rapid growth of realtime streaming applications. Many such applications play back stored audio and video over the network. However most of these applications do not perform effective congestion control, and there is significant concern about the effects on co-existing wellbehaved traffic and the potential for congestion collapse. In addition, most such applications are unable to perform quality adaptation on-the-fly as available bandwidth changes during a session, and so do not make effective use of additional bandwidth when it is available. This paper aims to provide some architectural insights on the design of video playback applications in the Internet. We identify end-to-end congestion control, quality adaptation and error control as the three major building blocks for Internet video playback

Present distributed applications are not typically designed to adapt themselves to changes in network conditions. In addition, network-based adaptation does not typically consider the requirements of specific applications and, therefore, may take actions contradictory to the application's needs. This paper presents a cooperative solution in which a configurable communication layer is used to adapt communication in response to both application requirements and network resource availability. Adaptation decisions are based on (1) payoff functions which capture the requirements of the application in a functional form, and (2) service availability curves which represent network resource availability. Experimentation with multimedia applications and a variable reliability protocol demonstrates the benefit of using payoff-based adaptation. 1 Motivation and Research Goals Interactive applications will continue to push the limits of available network bandwidths, processing speeds, and other co...

This paper presents a trace-driven simulation study of three classes of retransmission timeout (RTO) estimators in the context of low-bitrate real-time streaming over the Internet. We explore the viability of employing retransmission timeouts in NACK-based real-time streaming applications that support multiple retransmission attempts per lost packet. In such applications, real-time RTO estimation plays a major role (i.e., poor RTO estimation results in a larger number of duplicate packets and sometimes more frequent underflow events). Our study is based on trace data collected during a number of real-time streaming tests conducted between our dialup clients in all 50 states of the U.S. (including 653 major U.S. cities) and our backbone video server during a seven-month period. First, we define a generic performance measure for assessing the quality of hypothetical RTO estimators based on the samples of the roundtrip delay (RTT) recorded in the trace data. Second, using this performance measure, we evaluate the class of TCP-like estimators, find the most optimal estimator given our performance measure, and establish power laws that describe the tradeoff between the optimal number of duplicate packets and the optimal timeout waiting time. Third, we introduce a new class of RTO estimators based on delay jitter and show that they perform significantly better than TCP-like estimators in NACKbased applications. Finally, we gain a major insight into the RTT process by establishing which tuning parameters of an RTO estimator make it optimal given our performance measure and our experimental data, and give our explanation of the observed phenomena. I.