The deployment of IP Multicast has fostered the development of a suite of applications, collectively known as the MBone tools, for real-time multimedia conferencingover the Internet. Two of these tools --- nv from Xerox PARC and ivs from INRIA --- provide video transmission using softwarebased codecs. We describe a new video tool, vic, that extends the groundbreaking work of nv and ivs with a more flexible system architecture. This flexibility is characterized by network layer independence, support for hardware-based codecs, a conference coordination model, an extensible user interface, and support for diverse compression algorithms. We also propose a novel compression scheme called "IntraH. 261". Created as a hybrid of the nv and ivs codecs, IntraH. 261 provides a factor of 2-3 improvement in compression gain over the nv encoder (6 dB of PSNR) as well as a substantial improvement in run-time performance over the ivs H.261 coder. KEYWORDS Conferencing protocols; digital video; image ...

Most currently proposed solutions to application-level multicast organize the group members into an application-level mesh over which a DistanceVector routing protocol, or a similar algorithm, is used to construct source-rooted distribution trees. The use of a global routing protocol limits the scalability of these systems. Other proposed solutions that scale to larger numbers of receivers do so by restricting the multicast service model to be single-sourced. In this paper, we propose an application-level multicast scheme capable of scaling to large group sizes without restricting the service model to a single source. Our scheme builds on recent work on Content-Addressable Networks (CANs). Extending the CAN framework to support multicast comes at trivial additional cost and, because of the structured nature of CAN topologies, obviates the need for a multicast routing algorithm. Given the deployment of a distributed infrastructure such as a CAN, we believe our CAN-based multicast scheme offers the dual advantages of simplicity and scalability.

The current model for multicast transmission of video over the Internet assumes that a fixed average bandwidth is uniformly present throughout the network. Consequently, sources limit their transmission rates to accommodate the lowest bandwidth links, even though high-bandwidth connectivity might be available to many of the participants. We propose an architecture where a video transmission can be decomposed into multiple sessions with different bandwidth requirements using an application-level gateway. Our video gateway transparently connects pairs of sessions into a single logical conference by manipulating the data and control information of the video streams. In particular, the gateway performs bandwidth adaptation through transcoding and rate-control. We describe an efficient algorithm for transcoding Motion-JPEG to H.261 that runs in real-time on standard workstations. By making the Real-time Transport Protocol (RTP) an integral component of our architecture, the video gateway in...

In recent years, the "Internet Multicast Backbone," or MBone, has risen from a small, research curiosity to a largescale and widely used communications infrastructure. A driving force behind this growth was the development of multipoint audio, video, and shared whiteboard conferencing applications. Because these real-time media are transmitted at a uniform rate to all of the receivers in the network, a source must either run at the bottleneck rate or overload portions of its multicast distribution tree. We overcome this limitation by moving the burden of rate adaptation from the source to the receivers with a scheme we call receiver-driven layered multicast, or RLM. In RLM, a source distributes a hierarchical signal by striping the different layers across multiple multicast groups, and receivers adjust their reception rate by simply joining and leaving multicast groups. In this paper, we describe a layered video compression algorithm which, when combined with RLM, provides a comprehensive solution for scalable multicast video transmission in heterogeneous networks. In addition to a layered representation, our coder has low complexity (admitting an efficient software implementation) and high loss resilience (admitting robust operation in loosely controlled environments like the Internet) . Even with these constraints, our hybrid DCT/wavelet-based coder exhibits good compression performance. It outperforms all publicly available Internet video codecs while maintaining comparable run-time performance. We have implemented our coder in a "real" application---the UCB/LBL videoconferencing tool vic. Unlike previous work on layered video compression and transmission, we have built a fully operational system that is currently being deployed on a very large scale over the MBone.

Although IP Multicast is an effective network primitive for best-effort, large-scale, multi-point communication, many multicast applications such as shared whiteboards, multi-player games and software distribution require reliable data delivery. Building services like reliable sequenced delivery on top of IP Multicast has proven to be a hard problem. The enormous extent of network and end-system heterogeneity in multipoint communication exacerbates the design of scalable end-to-end reliable multicast protocols. In this paper, we propose a radical departure from the traditional end-to-end model for reliable multicast and instead propose a hybrid approach that leverages the successes of unicast reliability protocols such as TCP while retaining the efficiency of IP multicast for multi-point data delivery. Our approach splits a large heterogeneous reliable multicast session into a number of multicast data groups of co-located homogeneous participants. A collection of application-aware agents--Reliable Multicast proxies (RMXs)--organizes these data groups into a spanning tree using an overlay network of TCP connections. Sources transmit data to their local group, and the RNLX in that group forwards the data towards the rest of the data groups. RMXs use detailed knowledge of application semantics to adapt to the effects of heterogeneity in the environment. To demonstrate the efficacy of our architecture, we have built a prototype implementation that can be customized for different kinds of applications.

In just a few years the "Internet Multicast Backbone", or MBone, has risen from a small, research curiosity to a large scale and widely used communications infrastructure. A driving force behind this growth was our development of multipoint audio, video, and shared whiteboard conferencing applications that are now used daily by the large and growing MBone community. Because these real-time media are transmitted at a uniform rate to all the receivers in the network, the source must either run below the bottleneck rate or overload portions of the multicast distribution tree. In this dissertation, we propose a solution to this problem by moving the burden of rate-adaptation from the source to the receivers with a scheme we call Receiver-driven Layered Multicast, or RLM. In RLM, a source distr...

\Soft state " is an often cited yet vague concept in network protocol design in which two or more network entities intercommunicate in a loosely coupled, often anonymous fashion. Researchers often de ne this concept operationally (if at all) rather than analytically: a source of soft state transmits periodic \refresh messages " over a (lossy) communication channel to one or more receivers that maintain a copy of that state, which in turn \expires " if the periodic updates cease. Though a number of crucial Internet protocol building blocks are rooted in soft state-based designs | e.g., RSVP refresh messages, PIM membership updates, various routing protocol updates, RTCP control messages, directory services like SAP, and so forth | controversy is building as to whether the performance overhead of soft state refresh messages justify their qualitative bene t of enhanced system \robustness". We believe that this controversy has risen not from fundamental performance tradeo s but rather from our lack of a comprehensive understanding of soft state. To better understand these tradeo s, we propose herein a formal model for soft state communication based on a probabilistic delivery model with relaxed reliability. Using this model, we conduct queueing analysis and simulation to characterize the data consistency and performance tradeo s under a range of workloads and network loss rates. We then extend our model with feedback and show, through simulation, that adding feedback dramatically improves data consistency (by up to 55%) without increasing network resource consumption. Our model not only provides a foundation for understanding soft state, but also induces a new fundamental transport protocol based on probabilistic delivery. Toward this end, we sketch our design of the \Soft State Transport Protocol " (SSTP), which enjoys the robustness of soft state while retaining the performance bene t of hard state protocols like TCP through its judicious use of feedback. This research was supported by DARPA contract N66001-96-C-8508, by the State of California under the MICRO program, and by

Real-time multimedia streams like audio and video are now integral data types in modern programming environments. Although a great deal of research has investigated effective and efficient programming support for manipulating such streams and although the design of digital media "middleware" is fairly well understood, no widely available or commonly accepted programming model exists within the research community. We believe this lack of common practice impedes our collective progress because it prevents disparate research groups from easily leveraging each other's work. In this paper, we propose a solution to this problem that combines the best features of a number of existing multimedia toolkits --- Berkeley's Continuous Media Toolkit, MIT's VuSystem, and the LBL/UCB MBone tools --- into a fine-grained, extensible, and highperformance toolkit. We describe the convergence of these three toolkits into a common programming infrastructure and argue that the availability and acceptance of ...

Current Internet multicast conferencing tools treat all sources with equal importance in that they either statically allocate a fixed bandwidth to each source in a session, or they automatically adapt each source's transmission rate independently of all other sources. But not all sources are of equal interest to all receivers. We believe that to effectively support human to human communication, this disparity in receiver interest should be reflected in the rate-adaptation process. To this end, we propose a protocol called "SCUBA" that enables media sources to intelligently account for receiver interest in their rate-adjustment algorithms. SCUBA is orthogonal to and complements existing rate-adaptation schemes and can interoperate with either sender- or receiver-directed control systems. To scale the SCUBA protocol with multicast session size, we decouple the receiver-feedback process from the session size through sampling. This approach introduces a "tunable" tradeoff between convergen...

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...