Cutting audio latency with bidirectional WebRTC

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This talk will discuss the methods used to reduce latency in a bidirectional WebRTC pipeline, i.e., video but primarily audio in both directions. The primary topics are - Splitting the pipeline using intersrc/intersink, discussing the global pipeline latency and how to do avoid unnecessary clock syncing. This discusses pipeline latency and what is needed to allow media streaming in two directions. - Investigating different audio sinks, mainly pipewire and alsa and its effect on overall latency, both pipeline latency as well as perceived/measured latency. - Possible further improvements. Jitterbuffer, external properties (pipewire properties like quantum), mentioning webrtcbin2 from GStreamer that should potentially solve the first issue on its own. The end result reduced latency from 650-700ms to roughly 230ms, with as low as 180ms depending on the setup, making real-time communication possible, with a goal of 150ms.

Creation date: Oct. 24, 2025
Speakers: Albert Sjölund
License: CC BY-SA 3.0
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