New York, NY - Sep 12, 2013 - The Audio Engineering Society has published AES67-2013, a new engineering standard for networked/streaming audio-over-IP interoperability.
High-performance media networks support professional quality audio (16-bit, 44.1kHz 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 was initiated by the AES in December 2010 under the project name AES-X192. In August 2012, the AES and EBU jointly announced an active collaboration 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 to refine and clarify the necessary parameters.
The AES Standards Committee is the organization responsible for the standards program of the Audio Engineering Society. It develops and publishes a number of technical standards, information documents and technical reports. Working groups and task groups with a fully international membership are engaged in writing standards covering fields that include topics of specific relevance to professional audio. Membership of any AES standards working group is open to all individuals who are materially and directly affected by the documents that may be issued under the scope of that working group.