We describe a mechanism for scalable control of multicast continuous media streams. The mechanism uses a novel probing mechanism to solicit feedback information in a scalable manner and to estimate the number of receivers. In addition, it separates the congestion signal from the congestion control algorithm, so as to cope with heterogeneous networks. This mechanism has been implemented in the IVS videoconference system using options within RTP to elicit information about the quality of the video delivered to the receivers. The H.261 coder of IVS then uses this information to adjust its output rate, the goal being to maximize the perceptual quality of the image received at the destinations while minimizing the bandwidth used by the video transmission. We find that our prototype control mechanism is well suited to the Internet environment. Furthermore, it prevents video sources from creating congestion in the Internet. Experiments are underway to investigate how the scalable probing mech...

We use the measured round trip delays of small UDP probe packets sent at regular time intervals to characterize the end-to-end packet delay and loss behavior in the Internet. By varying the interval between probe packets, it is possible to study the structure of the Internet load over different time scales. In this paper, the time scales of interest range from a few milliseconds to a few minutes. Our observations agree with results obtained by others using simulation and experimental approaches. For example, our estimates of Internet workload are consistent with the hypothesis of a mix of bulk traffic with larger packet size, and interactive traffic with smaller packet size. The interarrival time distribution for Internet packets is consistent with an exponential distribution. We also observe a phenomenon of compression (or clustering) of the probe packets similar to the acknowledgement compression phenomenon recently observed in TCP. Our results also show interesting and less expected...

Network support for multicast has triggered the development of group communication applications such as multipoint data dissemination and multi-party conferencing tools. To support these applications, several multicast transport protocols have been proposed and implemented. Multicast transport protocols have been an area of active research for the past couple of years. This document tries to summarize the activities in this work-in-progress area by surveying several multicast transport protocols. The paper also presents a taxonomy to classify the surveyed protocols according to several distinct features, discusses the rationale behind the protocol's design decisions, and presents some current research issues in multicast protocol design. 1 Introduction Multicast transport mechanisms have been a topic of intense research and development efforts over the past couple of years. Both the Internet Engineering and Internet Research Task Forces (IETF and IRTF) have been heavily involved in co...

he video delivered to some of participants will be of low quality (since packet losses and/or delays will be high on some branches of the tree). The second case is safest, but it results in disturbing the bulk of the participants. In practice, it is not clear how to choose an adequate value for this fraction. In IVS, we set the fraction to a few percent [2, 3]. Ideally, however, the source should be able to single out the parts of the multicast tree that experience congestion. In order not to disturb the bulk of participants in the tree, these branches should be treated separately. Two possible solutions include video gateways, and using some form of layered coding. Video gateways or layered coding schemes are not new ideas. Our contribution is to identify and discuss the issues associated with using these techniques in the Internet. We illustrate our points with the H.261[10] software coder of IVS. 2 Video gateways Video gateways take

In this paper we address the problem of robust video transmission in error prone environments. The approach is compatible with the ITU-T video coding standard H.263. Fading situations in mobile networks are tolerated and the image quality degradation due to spatio-temporal error propagation is minimized utilizing a feedback channel between transmitter and receiver carrying acknowledgment information. In a first step, corrupted Group of Blocks (GOBs) are concealed to avoid annoying artifacts caused by decoding of an erroneous bit stream. The GOB and the corresponding frame number are reported to the transmitter via the back channel. The encoder evaluates the negative acknowledgments and reconstructs the spatial and temporal error propagation. A low complexity algorithm for real-time reconstruction of spatio-temporal error propagation is described in detail. Rapid error recovery is achieved by INTRA refreshing image regions (Macroblocks) bearing visible distortion. The feedback channel m...

For a reliable, flow-controlled multicast transport protocol to scale, it must avoid the feedback implosion problem [3], particularly if the protocol targets arbitrarily large multicast groups communicating over lossy networks. Most existing feedback control mechanisms based on probabilistic suppression address the feedback implosion problem by suppressing feedback using timers based on round-trip time (RTT) information. This approach requires that all receivers compute RTT to the data source. We present an algorithm whose major benefit derives from the fact that it does not need to compute RTT from receivers to the source, and does not require knowledge of group membership or network topology. We use a small set of representative receivers and probabilistic suppression to limit feedback. We believe that our approach will perform well in real networks. Simulations using randomly-generated network topologies of varying sizes with pessimistic network loss rates show that representatives ...

Distribution of continuous media traffic such as digital audio and video over packet-switching networks has become increasingly feasible due to a number of technology trends leading to powerful desktop computers and high-speed integrated services networks. Protocols supporting the transmission of continuous media are already available. In these protocols, transmission errors due to packet loss are generally not recovered. Instead existing protocol designs focus on preventive error control techniques that reduce the impact of losses by adding redundancy, e.g., forward error correction, or by preventing loss of important data, e.g., channel coding. The goal of this study is to show that retransmission of continuous media data often is, contrary to conventional wisdom, a viable option in most packet-switching networks. If timely retransmission can be performed with a high probability of success, a retransmission-based approach to error control is attractive because it imposes little overh...

Datagram networks such as the Internet do not provide guaranteed resources such as bandwidth or guaranteed performance measures such as maximum delay. One way to support packet video in these networks is to use feedback mechanisms that adapt the output rate of video coders based on the state of the network. In this paper, we present one such mechanism. We describe the feedback information, and how it is used by the coder control algorithm. We also examine how the need to operate in a multicast environment impacts the design of the control mechanism. Our mechanism has been implemented in the H.261 video coder of IVS. IVS is a videoconference system for the Internet developed at INRIA. Experiments indicate that the control mechanism is well suited to the Internet environment. In particular, it makes it possible to establish and maintain quality videoconferences even across congested connections in the Internet. Furthermore, it prevents video sources from swamping the resources of the Int...

We consider the problem of distributing audio data over networks such as the Internet that do not provide support for real-time applications. Experiments with such networks indicate that audio quality is mediocre in large part because of excessive audio packet losses. In this paper, we show using measurements over the Internet as well as analytic modeling that the number of consecutively lost audio packets is small unless the network load is very high. This indicates that open loop error control mechanisms based on forward error correction would be adequate to reconstruct most lost audio packets. .

The SHARE project seeks to apply information technologies in helping design teams gather, organize, re-access, and communicate both informal and formal design information to establish a "shared understanding" of the design and design process. This paper presents the visions of SHARE, along with the research and strategies undertaken to build an infrastructure toward its realization. A preliminary prototype environment is being used by designers working on a variety of industry sponsored design projects. This testbed continues to inform and guide the development of NoteMail, MovieMail, and Xshare, as well other components of the next generation SHARE environment that will help distributed design teams work together more effectively.