prlanzarin 0e162f1cda feat: configurable DSCP marking for WebRTC media ... RTCRTPSender exposes DSCP marking via `networkPriority` in the encodings configuration dictionaries. That should allow us to control QoS priorities for different media streams, eg audio with higher network priority than video. The only browser that implements that right now is Chromium. To use this, the public.app.media.networkPriorities configuration in settings.yml. Audio, camera and screenshare priorities can be controlled separately. For further info on the possible values, see: - https://www.w3.org/TR/webrtc-priority/ - https://datatracker.ietf.org/doc/html/rfc8837#section-5		2022-08-15 21:24:05 +00:00
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base.js	fix(audio): fix change output device error log	2022-05-03 14:51:30 +00:00
bridge-whitelist.js	TEMP	2021-12-09 20:37:05 +00:00
service.js	fix(audio): centralize default in/out device id definitions, make them an empty string	2022-04-11 19:23:32 +00:00
sfu-audio-bridge.js	feat: configurable DSCP marking for WebRTC media	2022-08-15 21:24:05 +00:00
sip.js	Merge pull request #15358 from frankemax/fix-audio-infinite-joining	2022-07-15 15:24:14 -03:00