Router mechanisms designed to achieve fair bandwidth allocations, like Fair Queueing, have many desirable properties for congestion control in the Internet. However, such mechanisms usually need to maintain state, manage buffers, and/or perform packet scheduling on a per flow basis, and this complexity may prevent them from being cost-effectively implemented and widely deployed. In this paper, we propose an architecture that significantly reduces this implementation complexity yet still achieves approximately fair bandwidth allocations. We apply this approach to an island of routers -- that is, a contiguous region of the network -- and we distinguish between edge routers and core routers. Edge routers maintain per flow state; they estimate the incoming rate of each flow and insert a label into each packet header based on this estimate. Core routers maintain no per flow state; they use FIFO packet scheduling augmented by a probabilistic dropping algorithm that uses the packet labels an...

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.

Streaming audio and video applications are becoming increasingly popular on the Internet, and the lack of effective congestion control in such applications is now a cause for significant concern. The problem is one of adapting the compression without requiring video-servers to re-encode the data, and fitting the resulting stream into the rapidly varying available bandwidth. At the same time, rapid fluctuations in quality will be disturbing to the users and so should be avoided. In this paper we present a mechanism for using layered video in the context of unicast congestion control. This quality adaptation mechanism adds and drops layers of the video stream to perform long-term coarse-grain adaptation, while using a TCP-friendly congestion control mechanism to react to congestion on very short timescales. The mismatches between the two timescales are absorbed using buffering at the receiver. We present a piecewiseoptimal scheme for the distribution of buffering among the active layers in order to maximize perceptual quality while minimizing rapid, disturbing changes in the quality. We discuss the issues involved in implementing and tuning such a mechanism, and present our simulation and experimental results.

Multicast (multipoint) distribution of video is an important component of many existing and future networked services. Today's Internet lacks support for quality of service (QoS) assurance which makes the transmission of real-time traffic (such as video) challenging. In addition, the heterogeneity of the Internet's transmission resources and end-systems makes it extremely difficult, if not impossible, to agree on acceptable traffic characteristics among multiple receivers of the same video stream. In this paper we survey techniques that have been proposed for transmitting video in this environment. These techniques generally involve adaptation of the video traffic carried over the network to match receiver requirements and network conditions. In addition to their applicability to the near-term capabilities of the Internet, they also are of relevance to a future, QoS-aware Internet environment because of the inevitable inaccuracies in traffic and resource reservation specifica...

Router mechanisms designed to achieve fair bandwidth allocations, like Fair Queueing, have many desirable properties for congestion control in the Internet. However, such mechanisms usually need to maintain state, manage buffers, and/or perform packet scheduling on a per flow basis, and this complexity may prevent them from being cost-effectively implemented and widely deployed. In this paper, we propose an architecture that significantly reduces this implementation complexity yet still achieves approximately fair bandwidth allocations. We apply this approach to an island of routers -- that is, a contiguous region of the network -- and we distinguish between edge routers and core routers. Edge routers maintain per flow state; they estimate the incoming rate of each flow and insert a label into each packet header based on this estimate. Core routers maintain no per flow state; they use FIFO packet scheduling augmented by a probabilistic dropping algorithm that uses the packet labels and an estimate of the aggregate traffic at the router. We call the scheme Core-Stateless Fair Queueing. We present simulations and analysis on the performance of this approach.

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 ...

Streaming audio and video applications are becoming increasingly popular on the Internet, and the lack of effective congestion control in such applications is now a cause for significant concern. The problem is one of adapting the compression without requiring video-servers to reencode the data, and fitting the resulting stream into the rapidly varying available bandwidth. At the same time, rapid fluctuations in quality will be disturbing to the users and should be avoided. In this paper we present a mechanism for using layered video in the context of unicast congestion control. This quality adaptation mechanism adds and drops layers of the video stream to perform long-term coarse-grain adaptation, while using a TCP-friendly congestion control mechanism to react to congestion on very short timescales. The mismatches between the two timescales are absorbed using buffering at the receiver. We present an efficient scheme for the distribution of available bandwidth among the active layers. Our scheme allows the server to trade short-term improvement for long-term smoothing of quality. We discuss the issues involved in implementing and tuning such a mechanism, and present our simulation results. Keywords--- Quality Adaptive Video Playback, Unicast Layered Transmission, Internet I.

In this paper, we present an ecient Multiple Description Coding (MDC) technique to achieve robust communication over unreliable channels such as a lossy packet network. We first model such unreliable channels as erasure channels and then we present a MDC system using polyphase transform and selective quantization to recover channel erasures. Different from previous MDC work, our system explicitly separates description generation and redundancy addition which greatly reduces the implementation complexity specially for systems with more than two descriptions. Our system also realizes a Balanced Multiple Description Coding (BMDC) framework which can generate descriptions of statistically equal rate and importance. This property is well matched to communication systems with no priority mechanisms for data delivery, such as today's Internet.

Today’s Internet provides one simple service: best effort datagram delivery. This minimalist service allows the Internet to be stateless, that is, routers do not need to maintain any fine grained information about traffic. As a result of this stateless architecture, the Internet is both highly scalable and robust. However, as the Internet evolves into a global commercial infrastructure that is expected to support a plethora of new applications such as IP telephony, interactive TV, and e-commerce, the existing best effort service will no longer be sufficient. In consequence, there is an urgent need to provide more powerful services such as guaranteed services, differentiated services, and flow protection. Over the past decade, there has been intense research toward achieving this goal. Two classes of solutions have been proposed: those maintaining the stateless property of the original Internet (e.g., Differentiated Services), and those requiring a new stateful architecture (e.g., Integrated Services). While stateful solutions can provide more powerful and flexible services such as per flow bandwidth and delay guarantees, they are less scalable than stateless solutions. In particular, stateful solutions require each router to maintain and manage per flow state on the control path, and to perform per flow classification, scheduling, and buffer management on the data path. Since today’s routers can

We describe a framework for video transmission over the Internet that features the coordinated operation of an application-layer video source coding algorithm and a transport-layer rate control mechanism. The proposed video coding scheme operates on a progressively encoded video stream and provides graceful resilience to network packet drops. The robustness is enabled through a generalized Multiple Description (MD) coding strategy, architected as an adaptive array of packet-erasure correction codes. The video coding algorithm is matched to an efficient and reactive rate control mechanism that minimizes the fluctuation of rate and uses the profile of past losses to adjust the rate in a TCP-friendly manner. While the two constituent algorithms identified above are interesting in their own right, a key feature of this work is the integration of these algorithms in a simple framework that seeks to maximize the expected delivered video quality at the receiver through coordinated adaptation...