Thanks for your kind words -- you're certainly welcome to contact us if you think we can help. Yes, as I wrote elsewhere here, it would probably be possible to do everything Salsify does within the context of WebRTC, if you get to change the sender and receiver. If you're only changing the sender and you want to interoperate with receivers running the WebRTC.org codebase (e.g. Chrome), you have less flexibility. If you're only changing the sender and you want to work with existing high-performance video codecs and the API they expose, even less flexibility.

My main advice for implementers would be, benchmark your end-to-end glass-to-glass video latency (including the time spent waiting for the next frame to be captured & encoded, and then the end-to-end latency through to the display of that frame) over varied/unpredictable networks. In my experience, implementers sometimes get caught up focusing too much on network-layer measurements (IP latency) and can end up missing what in my view is the bottom line: glass-to-glass video latency. You can use our mahimahi mm-delay/mm-link/mm-onoff/mm-loss tools (part of Debian/Ubuntu) and the included traces to model some bad networks. And then see how you do on the same traces we use in our paper. If you can make a plot like our Figure 8(a) and it all looks good, that seems like good progress in my book.