5 ms·
Only superficially familiar with WebRTC - what does this thing do to get the claimed 'low latency'?
by jarym 6y ago
Only superficially familiar with WebRTC - what does this thing do to get the claimed 'low latency'?
- jtsiskin 6y agoI don’t consider 500ms “low latency”. You should be below 250ms before you start touting it as a key selling point.
- dindresto 6y agoLive streaming video usually has a latency of 6 to 30 seconds, so in that context anything below a second is very low latency Edit: the topic of my bachelor's thesis included low latency live streaming, so I hope I have at least some knowledge :)
- thakoppno 6y agothis application is for real-time communication more so than live streaming.
- sippingjippers 6y agoYou are extremely lucky if you get a contemporary online livestream under 10 seconds, this includes AFAIK all of YouTube, and the vast majority of HLS-based streams - which are often 30 seconds latency or worse. Getting buffers down to 500ms is already pretty insane assuming reliability doesn't suffer. 250ms is well within the envelope of a single drop on a 3G network killing playback
- CyberDildonics 6y agoThere are plenty of twitch streams that have latency under 2 seconds in low latency mode. You have to watch the source stream and the streamer can't use larger buffers, but it has already been done for well over a year or more. If a streamer has the chat window on screen you can type something in and see it on stream in two seconds. I have seen private mixer streams have 2 second latency also.
- hmtdmr 6y agoAccording to many sources, sub-second is considered to be ultra low latency. 500ms is good for now but 250ms would be great.
- CharlesW 6y agoI'm interested in this too, and why the "low latency" mode is enabled in everything but the community version.
- noen 6y agoQuite simply, this isn't low latency. The way you drop latency is by reducing or eliminating buffers, that's pretty much it. There's a trade-off with h264 and hevc/265, as with most video codecs - the larger the buffer you have, the more delta frames you can have between key frames. This means lower bandwidth and better compression. So it's a constant struggle between compression and latency. The lowest latency that webrtc can get is about 4ms one way, 8ms round trip, at least based on the project I started to do realtime steaming with https://github.com/3DStreamingToolkit/3DStreamingToolkit https://github.com/3DStreamingToolkit/3DStreamingToolkit We proved you could do round trip times in the real world from cloud to customer in under 25ms. But it comes with a lot of tradeoffs and cost and complexity to make it work. Google Stadia fundamentally uses webrtc under the covers, although they use QUIC instead of TCP/ICE. If you ripped out all of the congestion management from the video engine in webrtc, you'd reduce another several ms roundtrip at the cost of basically no network resilience.
- dep_b 6y agoWebRTC is low latency by itself, otherwise you couldn’t have a video call. The challenge is to serve one WebRTC stream to 1,000,000 viewers while maintaining low latency