WebRTC Fine Tuning #239

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opened 2026-10-09 08:44:34 -04:00 by rainbownapkin · 1 comment

rainbownapkin commented

2026-10-09 08:44:34 -04:00

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We should adjust the JitterBufferTarget value to around a second or two to improve stream consistency as our use-case isn't very latency sensitive.

Furthermore, we should also cap stream bitrates to keep far away from monthly bandwidth caps from the VPS provider that hosts our TURNS infrastructure.

We should adjust the [JitterBufferTarget](https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpReceiver/jitterBufferTarget) value to around a second or two to improve stream consistency as our use-case isn't very latency sensitive. Furthermore, we should also cap stream bitrates to keep far away from monthly bandwidth caps from the VPS provider that hosts our TURNS infrastructure.

rainbownapkin added this to the 0.3-Alpha (Acapulco Gold) milestone 2026-10-09 08:44:34 -04:00

rainbownapkin added the

Performance Improvement

UX/Accessibility

labels 2026-10-09 08:44:34 -04:00

rainbownapkin self-assigned this 2026-10-09 08:44:34 -04:00

rainbownapkin commented

2026-10-10 05:10:14 -04:00

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Owner

Changed webRTC jitterbuffer to 1.5s for clients receiving streams, updated WebRTC text to reflect recent TURN(S) capabilities - 881dcc2e9c

Capped bitrates for WebRTC Streams - 1dfe3454b4

Disabled unnecessary audio processing for webRTC Streams - d686cf540e

Changed webRTC jitterbuffer to 1.5s for clients receiving streams, updated WebRTC text to reflect recent TURN(S) capabilities - 881dcc2e9c Capped bitrates for WebRTC Streams - 1dfe3454b4 Disabled unnecessary audio processing for webRTC Streams - d686cf540e

rainbownapkin closed this issue

2026-10-10 05:10:14 -04:00