We describe Service Fabric (SF), Microsoft's distributed platform for building, running, and maintaining microservice applications in the cloud. SF has been running in production for 10+ years, powering many critical services at Microsoft. This paper outlines key design philosophies in SF. We then adopt a bottom-up approach to describe low-level components in its architecture, focusing on modular use and support for strong semantics like fault-tolerance and consistency within each component of SF. We discuss lessons learned, and present experimental results from production data.

Retransmission can be used for loss recovery in continuous media applications but the number of retransmission attempts is bounded by the size of the playout buffer. For efficient recovery, a protocol must attempt as many retransmissions as possible but avoid late retransmissions. This typically requires that the playout buffer be sized in round-trip time (RTT) multiples plus some margin for error. RTT-based timers are then used to trigger retransmissions. However, this approach is problematic due to (i) the high variation in RTT commonly encountered in the Internet, and (ii) the coarse granularity of timers typically used in protocol implementations.We present two new retransmission-based protocols, for unicast and multicast respectively, which eliminate RTT estimation and timer-triggered events. As a result, our protocols are immune to errors due to RTT jitter and timer granularity and recover more losses, while better suppressing unnecessary retransmission requests and retransmissions than timer-based protocols. At the same time, our protocols are simpler to implement and degrade more gracefully than timer-based protocols.

With the advent of high-speed networks such as Internet2, high quality uncompressed transmission of multi-channel audio streams has become possible. For interactive applications, such as a distributed musical performance, minimizing latency is of paramount importance. Given the strict latency requirements, error recovery (via either retransmission or FEC) may not always be successful, and thus concealment is frequently required. In this paper we propose a novel concealment algorithm, based on inter-channel redundancy, for multi-channel, professional quality, uncompressed audio streams, with particular emphasis on an experimental 10.2 audio standard, which provides an immersive experience for the audience and the players in a performance. We also propose a smoothing method based on Bezier curves. We focus on interactive applications; thus we investigate concealment techniques that can be performed in real time. Our algorithms are implemented in a testbed capable of streaming up to 24 uncompressed audio channels with end-to-end latency of less than 6 ms. Our results show that our techniques outperform existing methods. We expect that our protocol will become an important part of a distributed immersive musical performance system currently being developed at our university.