AES67 is a standard for high-quality audio over IP (Internet Protocol) networks, developed by the Audio Engineering Society (AES). The standard defines a set of protocols and specifications for transmitting audio over IP networks in a way that is compatible with existing professional audio equipment.
AES67 is based on the RTP (Real-time Transport Protocol) and uses the same packet structure as other RTP-based protocols such as SIP, RTCP, and SRTP. It also uses the same audio codecs as other RTP-based protocols, such as PCM and compressed formats like AAC and Opus.
One of the key features of AES67 is its support for low-latency audio transmission, which is essential for professional audio applications such as live sound reinforcement, broadcast, and studio recording. The standard specifies a maximum end-to-end latency of less than 10 milliseconds for uncompressed audio, and less than 20 milliseconds for compressed audio.
AES67 also supports multicasting, which allows for the efficient transmission of audio to multiple destinations simultaneously. This is useful for applications such as large-scale live events where audio needs to be distributed to multiple locations.
In terms of network compatibility, AES67 is designed to work on standard Ethernet networks and can be used over both wired and wireless connections. It can also be used over existing network infrastructures, including existing IPv4 and IPv6 networks, and can be integrated with other network protocols such as Dante, Livewire, and Q-LAN.
In summary, AES67 is a professional audio over IP standard that provides high-quality, low-latency audio transmission over standard Ethernet networks. It's compatible with existing professional audio equipment, and can be integrated with other network protocols. It supports multicasting and is designed to work on standard Ethernet networks and can be used over both wired and wireless connections.