Native WebRTC provides methods to increase latency on the remote sources. Increased latency opens room for additional optimizations and better audio, video quality. We intend to create an origin trial that enables users to set this value, and read this value in javascript layer.


Specification link

Specification being incubated in a Community Group

Status in Chromium


No active development

Consensus & Standardization

After a feature ships in Chrome, the values listed here are not guaranteed to be up to date.

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Intent to Prototype url

Intent to Prototype thread

Search tags

WebRTC, JitterBuffer, Blink, Javascript, audio, video,

Last updated on 2020-11-19