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Audio-over-IP (Dante)

Pick channels, sample rate (48/96/192 kHz) and bit depth and I compute the raw audio bandwidth and the network estimate (with RTP/IP overhead), packets per second and the Dante latency.

ch

Total simultaneous audio channels. Each PCM channel takes Fs × bits per second.

48 kHz is the professional standard. Doubling the sample rate doubles the bandwidth.

Dante carries PCM. 24 bit is the norm in production and live sound.

Lower latency = more packets per second (more CPU/network load). 1 ms is the Dante default.

Scenarios
redzilla.cl — dante
 
Network BW · estimate
with RTP/IP overhead
Audio BW · raw
PCM, no headers
Packets / sec
per flow
Samples / packet
per channel
Raw
Network
Pkt/s
Latency

1 GbE link usage

1000 Mbps

Network estimate against a Gigabit Ethernet link (1000 Mbps).

Bandwidth by channels

ChannelsRawNetwork% of 1 GbE

current setup · at the chosen sample rate and bit depth.

How it is calculated · audio over IP

1. Raw audio bandwidth (PCM samples only): BW = channels × Fs × bits, with Fs in Hz. E.g. 1 × 48000 × 24 = 1.152 Mbps per channel at 48 kHz/24 bit.

2. Network bandwidth: we multiply the raw figure by an overhead factor (×1.3 by default) that approximates the RTP 12 + UDP 8 + IP 20 + Ethernet headers. The real value depends on packing (samples per packet): more latency means more samples per packet and lower relative overhead.

3. Packets per second per flow: pps = 1000 / latency_ms. At 1 ms that is 1000 packets/s; at 0.25 ms, 4000.

4. Samples per packet (per channel): Fs × latency_ms / 1000. At 48 kHz and 1 ms that is 48 samples.

5. This is an estimate for sizing the network. Dante and AES67 pack each flow somewhat differently and add control traffic (mDNS, PTP for clock sync). The fixed per-packet overhead weighs more with few samples per packet (low latencies) and few channels per flow.

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