The networking of live audio for professional applications typically uses layer 2-based solutions such as AES50 and MADI utilizing fixed time slots similar to Time Division Multiplexing (TDM). However, these solutions are not effective for best effort traffic where data traffic utilizes available bandwidth and is consequently subject to variations in QoS. There are audio networking methods such as AES47, which is based on asynchronous transfer mode (ATM), but ATM equipment is rarely available. Audio can also be sent over Internet Protocol (IP), but the size of the packet headers and the difficulty of keeping latency within acceptable limits make it unsuitable for many applications. In this paper we propose a new unified low latency network architecture that supports both time deterministic and best effort traffic toward full bandwidth utilization with high performance routing/switching. For live audio, this network architecture allows low latency as well as the flexibility to support multiplexing multiple channels with different sampling rates and word lengths.
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