5
AES67 Features
AES67
The AES67 standard defines an interoperability mode for transport of high-performance audio over networks
based on the Internet Protocol. For the purposes of the standard, high-performance audio refers to audio with full
bandwidth and low noise. These requirements imply linear PCM coding with a sampling frequency of 44.1 KHz
and higher, and resolution of 16-bits and higher. High performance also implies a low-latency capability compati-
ble with live sound applications. The standard considers latency performance of 10 milliseconds (ms) or less.
The xNode implementation of AES67 is compliant with the standard, supporting interoperability with third party
devices. In particular, the following features are implemented:
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Multicast and Unicast Real Time Protocol (RTP) audio streaming
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Session Initiation Protocol (SIP) based connection management for Unicast streams
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Transmission of 1ms audio frames in linear PCM 24-bit format
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Ability to receive audio streams containing 1 through 8 channels
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Ability to receive 16-bit as well as 24-bit format
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Support of IEEE1588-2001 Precision Time Protocol standard
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Time-stamping of outgoing streams with IEEE1588 derived RTP timestamps
Network Synchronization Setup
The AES67 standard requires devices to use the IEEE1588 network time protocol standard. Synchronization
related options are available on the QoS WEB page.
Figure 35 - AES67 Synchronization configuration
Summary of Contents for Linear Acoustic SDI XNODE
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