Packetization Process
Digital audio is divided into packets for IP network transport.
Each packet contains a fixed number of audio samples plus network headers (Ethernet, IP, UDP, RTP).
Latency Components
Latency comes from the time required to fill a packet (packetization),
stabilization buffers, and network transit. Larger packets increase latency.
Performance Trade-offs
Few samples per packet = low latency but many packets (network load).
Many samples = fewer packets but high latency.
Protocol Comparison
AES67 is an open standard using RTP (12 additional header bytes).
Dante is proprietary, based directly on UDP, slightly more efficient.
Bandwidth Calculation
Bandwidth = Total packet size × Packets per second.
Includes all network headers (overhead). An 8-channel stream @ 48kHz typically uses 5-10 Mbps.
Jitter Buffer Function
Reception buffer that compensates for variations in network delay (jitter).
Too low = dropouts. Too high = unnecessary latency. 2-5 ms sufficient on optimized networks.
AES67 Packet Time
AES67 defines packet times: 125µs, 250µs, 1ms (mandatory), 4ms.
At 48kHz: 1ms = 48 samples. This tool uses sample count for flexibility across both protocols.
Transport Modes
Unicast (1-to-1): Lower switch load, point-to-point delivery.
Multicast (1-to-many): Efficient for multiple receivers. AES67 primarily uses multicast.
Professional IP Audio Packetization Analysis Tool — Version 2.0
Support Development