We have a self hosted Wowza Streaming Engine. We’ve built an application that you can think of as “just another web conferencing tool” ;-). WebRTC is published from a browser to the WSE, and that stream is then consumed over WebRTC by a different browser.
We’re seeing browser-to-browser latency of about 1 to 2 seconds. That’s ok - but we want it to go faster.
I’m aware of a number of optimisations within the WSE for buffer size and flushing - to reduce the latency of (for example) transcoding. However, I can’t find any reference to similar optimisations for WebRTC.
So firstly, can I optimise the buffer for WebRTC? and if so, how?
And secondly, are there any other optimisation suggestions for this workflow?