State of the art, real-time, rate-adaptive, multimedia applications adjust their transmission rate to match the available network capacity. Unfortunately, this source-based rate-adaptation performs poorly in a heterogeneous multicast environment because there is no single target rate --- the conflicting bandwidth requirements of all receivers cannot be simultaneously satisfied with one transmission rate. If the burden of rate-adaption is moved from the source to the receivers, heterogeneity is accommodated. One approach to receiver-driven adaptation is to combine a layered source coding algorithm with a layered transmission system. By selectively forwarding subsets of layers at constrained network links, each user receives the best quality signal that the network can deliver. We and others have proposed that selective-forwarding be carried out using multiple IP-Multicast groups where each receiver specifies its level of subscription by joining a subset of the groups. In this paper, we ...

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 ...

Several recent proposals for an "active networks" architecture advocate the placement of user-defined computation within the network as a key mechanism to enable a wide range of new applications and protocols, including reliable multicast transports, mechanisms to foil denial of service attacks, intra-network real-time signal transcoding, and so forth. This laudable goal, however, creates a number of very difficult research problems, and although a number of pioneering research efforts in active networks have solved some of the preliminary small-scale problems, a large number of wide open problems remain. In this paper, we propose an alternative to active networks that addresses a restricted and more tractable subset of the active-networks design space. Our approach, which we (and others) call "active services", advocates the placement of user-defined computation within the network as with active networks, but unlike active networks preserves all of the routing and forwarding semantics o...

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.

We describe a mechanism for dynamic adjustment of the bandwidth requirements of multimedia applications. The sending application uses RTP receiver reports to compute packet loss and jitter. Based on these metrics the congestion state seen by the receivers is determined and the bandwidth is adjusted by a linear regulator with dead zone. The suggested mechanism has been implemented and controls the bandwidth of the vic video conferencing system. Currently we are evaluating the proposed algorithms by simulations and experiments on the Internet and on our local ATM network. 1 Introduction Multimedia applications pose unique challenges for network control and management: they offer high data rates, stringent real-time contraints, long connection durations and relatively inflexible demands on bandwidth. Due to the long connection duration, the standard cycle of QoS negotiation purely at the beginning of a session leads either to high call rejection probability at busy times or unnece...

\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

Multirate multicasting, where the receivers of a multicast group can receive service at different rates, is an efficient mode of data delivery for many real-time applications. In this paper, we address the problem of achieving rates that maximize the total receiver utility for multirate multicast sessions. This problem not only takes into account the heterogeneity in user requirements, but also provides a unified framework for diverse fairness objectives. We propose two algorithms and prove that they converge to the optimal rates for this problem. The algorithms are distributed and scalable, and do not require the network to know the receiver utilities. We discuss how these algorithms can be implemented in a real network, and also demonstrate their convergence through simulation experiments.

IP Multicast has proven to be an effective communication primitive for best effort, large-scale, multi-point audio/video conferencing applications. While the best-effort transport of real-time digital audio/video is a relatively straightforward and well understood problem, many other applications like multicast-based shared whiteboards and shared text editors are more challenging to design because their underlying media require reliable transport, i.e., a "reliable multicast" protocol. The design of scalable end-to-end reliable multicast protocols has unfortunately proven to be an especially hard problem, exacerbated by the enormous degree of network and system heterogeneity present in the Internet. In this paper, we propose to tackle the heterogeneity problem with a hybrid model for reliable multicast that relies in part on end-to-end loss recovery mechanisms and in part on intelligent and application-aware adaptation carried out within the network. In our framework, application-aware agents -- or proxies -- use detailed knowledge of application semantics to hide the effects of heterogeneity from the rest of the system. We present a general architecture for proxy-based reliable multicast called the Reliable Multicast proXy (RMX) model and describe a prototype implementation of an RMX for a shared whiteboard application for hand-held PDAs.

Controlling the rate of bulk data multicast to a large number of receivers is difficult, due to the heterogeneity among the end-systems' capabilities and their available network bandwidth. If the data transfer rate is too high, some receivers will lose data, and retransmissions will be required. If the data transfer rate is too slow, an inordinate amount of time will be required to transfer the data. In this paper, we examine an approach towards rate-controlled multicast of bulk data in which the sender uses multiple multicast groups to transmit data at different rates to different sub-groups of receivers. We present simple algorithms for determining the transmission rate associated with each multicast channel, based on static resource constraints, e.g., network bandwidth bottlenecks. Transmission rates are chosen so as to minimize the average time needed to transfer data to all receivers. Analysis and simulation are used to show that our policies for rate selection perform well for la...