New York, NY (September 16, 2013)—The Audio Engineering Society has published the AES67-2013, a new engineering standard for networked/streaming audio-over-IP interoperability.
High-performance media networks support professional quality audio (16 bit, 44.1 kHz and higher) with low latencies (less than 10 milliseconds) compatible with live sound reinforcement. The level of network performance required to meet these requirements is available on local-area networks and is achievable on enterprise-scale networks. A number of networked audio systems have been developed to support high-performance media networking, but until, now there were no recommendations for operating these systems in an interoperable manner. This standard provides comprehensive interoperability recommendations in the areas of synchronization, media clock identification, network transport, encoding and streaming, session description and connection management.
The project began in 2010, and it was formally announced in 2012 that the AES and EBU were collaborating to achieve interoperability of networked audio. The intent was not to invent new technology, but to identify an interoperable subset of existing technologies to achieve this goal. Task Group SC-02-12-H, under the leadership of Kevin Gross, met regularly using web conferencing and email to refine and clarify the necessary parameters.
To obtain a copy of this standard, go to http://www.aes.org/publications/standards/search.cfm?docID=96
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