Bandwidth is the maximum rate at which data can be transmitted over a network path, measured in bits per second.
In WebRTC, available bandwidth determines the quality of audio and video that can be maintained during a session. The BWE (Bandwidth Estimation) algorithm continuously probes and estimates the available bandwidth.
Key bandwidth concepts in WebRTC
- Dynamic fluctuation: Available bandwidth changes constantly as network conditions shift
- Asymmetric: Upload and download bandwidth are often different (especially on residential connections and mobile networks)
- Shared: WebRTC competes with other traffic on the same network path
- Bitrate vs bandwidth: Bitrate is the number of bits you need to transmit your compressed media; bandwidth is how fast the network can carry it. Encoding faster than available bandwidth causes congestion
- CBR vs VBR: Do you encode and send media at a constant bitrate or a variable one that depends on the content
Bandwidth allocation
When multiple media streams share a connection, WebRTC must allocate bandwidth between them:
- Audio is prioritized over video (audio uses little bandwidth but is critical for communication)
- Active speaker video may get more bandwidth than thumbnail views
- Simulcast allows the SFU to select appropriate quality per recipient
- Videos are prioritised based on their importance (see Bandwidth Allocation)
