What is the latency of webrtc in particular RTCMultiConnection?

Question

WebRTC

Someone tried to measure what kind of delay the browser gives when creating a conference in WebRTC in addition to network Ping.
As I understand it, there are codec delays, and while some buffer is being collected for transfer, and so on.

Answer 1

The delay in general depends on a lot of things
1) The jitter buffer has its own size and its own packet drop algorithms
2) The server through which the traffic goes (TURN or another server) can have its own buffers
The encoding and decoding delay of the stream is invisible to the eye.
Well, of course it depends on RTT.
How much delay do you have from RTT and how much do you want to improve it?

Answer 2

WebRTC supports RTCP, there is a lot of useful information there, you can also measure the delay.

Answer 3

I believe the author is asking about the delay before the video goes. The latency is high and this is an internal, fatal flaw in the SDP protocol that looks for ports available for communication. It may be possible to do some lowlevel hack to fix this problem, but it will be hard. I observed delays up to 20 seconds...