Layered multicast is a promising technique for broadcasting adaptive-quality TV video to heterogeneous receivers. While several-layered multicast approaches have been proposed, prior work has identified several problems including significant and persistent instability in video quality, arbitrary unfairness with other sessions, low access link utilization due to conservative bandwidth allocation, and problems with receiver synchronization. In this paper we propose a new layered multicast scheme, where we exploit a simple, coarse-grained, two-tier loss differentiation architecture to achieve stable and fair bandwidth allocation for viewers. Despite the simplicity of our loss differentiation model, we show that it achieves most of the benefits of complex and costly priority dropping schemes. In addition, our protocol is receiver-driven and thus retains the incentives to limit bandwidth usage that are not present in existing priority dropping schemes.

Packet telephony is of increasing interest in both the telecommunications and Internet communities. The emergence of packet telephony will create new services, and presents an opportunity to rethink how conventional telephony services are implemented. In this paper, we present an architecture for telephony over packet networks (TOPS). TOPS allows users to move between terminals or to use mobile terminals while being reachable by the same name. TOPS users can have multiple terminals and control how calls are routed to them. TOPS allows for terminals with a range of capabilities such as support for video, whiteboard, and other media with a variety of coding formats. TOPS retains the necessary information on terminal capabilities to determine the appropriate type of communication to be established with the remote terminal. The architecture assumes that the underlying network supports the establishment of end-to-end connectivity between terminals, with an appropriate quality of service. The components of TOPS are a directory service, an application layer signaling protocol, and a logical channel abstraction for communication between end-systems. The directory service maps a user's name to a set of terminals where the user may be reached. A user can control the translation operation by specifying profiles that customize how his name is mapped to a set of terminals where he can be reached. Terminal capabilities are also stored in the directory service. The application layer signaling protocol establishes and maintains call state between communicating terminals. The logical channel abstraction provides a shared end-to-end context for a call's constituent media and control streams, while isolating the applications from the details of the network transport mechanisms. In addition to supporting simple point-to-point calls, the architecture supports both centralized and decentralized conferencing. We also introduce a simple encapsulation format for voice.

Two important requirements for protocol implementations to be able to provide quality of service (QoS) guarantees within the endsystem are: (1) efficient processor scheduling for application and protocol processing and (2) efficient mechanisms for data movement. Scheduling is needed to guarantee that the application and protocol tasks involved in processing each stream execute in a timely manner and obtain their required share of the CPU. We have designed and implemented an operating system (OS) mechanism called the real-time upcall (RTU) to provide such guarantees to applications. The RTU mechanism provides a simple real-time concurrency model and has minimal overheads for concurrency control and context switching compared to thread-based approaches. To demonstrate its efficacy, we have built RTU-based transmission control protocol (TCP) and user datagram protocol (UDP) protocol implementations that combine high efficiency with guaranteed performance. For efficient data movement, we have implemented a number of techniques such as: (1) direct movement of data between the application and the network adapter; (2) batching of input-output (I/O) operations to reduce context switches; and (3) header-data splitting at the receiver to keep bulk data page aligned. Our RTU-based user-space TCP/Internet protocol (TCP/IP) implementation provides bandwidth guarantees for bulk data connections even with real-time and "best-effort" load competing for CPU on the endsystem. Maximum achievable throughput is higher than the NetBSD kernel implementation due to efficient data movement. Sporadic and small messages with low delay requirements are also supported using reactive RTUs that are scheduled with very low delay. We believe that ours is the first solution that combines good data path performance with application-level bandwidth and delay guarantees for standard protocols and OSs.

This paper seeks to bridge the gap between theory and practice of real-time scheduling in the domain of high speed multimedia networking. We show that the strict preemptive nature of real-time scheduling leads to more context switching, and requires system calls for concurrency control. We present our scheduling scheme called rate-monotonic with delayed preemption (rmdp) and show how it reduces both these overheads. We then develop the analytical framework to analyze rmdp and other scheduling schemes that lie in the region between strict (immediate) preemption and no preemption. Our idealized scheduler simulation methodology accounts for the blocking introduced by these schemes under the usual assumption that the time for context switching and preemption is zero. We derive simpler schedulability tests for non-preemptive scheduling, and prove a variant of rate-monotonic scheduling that has fewer preemptions. Our measurements on Sparc and Pentium platforms, show that for the workloads we considered, Rmdp increases useful utilization by as much as 8%. Thus our scheduling policies have the potential to improve performance over existing methods.

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