Is it possible to buffer the video/audio in WebRTC (of course, having then a delay on the other side) to improve the quality?
可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试):
问题:
回答1:
WebRtc does buffering automatically when it is necessary. You don't have to think about it.
回答2:
There's no exact way to fully control this, but there are some settings in the Opus codec that can influence it, like, minptime, ptime, maxptime. Please check https://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03#page-12 for more info.
