[imtringued]

My WebRTC projects haven't suffered that much from latency. The biggest source of delays is usually caused by encoding video for me. I've had to limit streams to 720p and 25fps to reduce the time spent on CPU encoding a vp8 stream. There are also bandwidth considerations (real time encoding = significantly less compression) but the end result is slightly less than 200ms one way latency (including input lag from mouse, 15ms network latency and display lag) without any special settings. All I'm doing is feeding a ffmpeg stream to kurento and letting it broadcast it via WebRTC. This is not a web conferencing application and it is also not using WebRTC via p2p. It's closer to conventional live streaming with a sane amount of latency (compared to up to 30s of latency you commonly see on twitch). Of course I personally would prefer it if the latency can be brought down even further. 100ms or lower is like the holy grail for me and only appears to be doable with codecs that aren't supported by WebRTC. However, people don't want to install apps just for my little service and I certainly won't encode every stream via several codecs just for the tiny minority of the user base that actually ends up using the app.