I don't think that means that a Salsify sender would be interoperable with, like, a Chrome receiver, though (even though we're just using VP8) -- we'd have to implement at least the receiver side of Salsify's congestion-control protocol inside WebRTC.org/Chrome, and Salsify sometimes likes to encode a VP8 frame that has to be interpreted relative to a certain (prior) decoder state.
On #2, honestly I've worked with libopus a bit for Puffer and it's just really pleasant to work with. As you say, a lot of the difficulties of interacting with a video encoder you just don't have in this context. It's also pretty easy to get "gapless/clickless" back-to-back playback of audio excerpts that were encoded completely independently, unlike with video where this is a huge pain in the neck and usually requires a SAP/IDR/closed GOP/sequence header+I-frame+P-frame (which takes a lot of bytes so you can't do it very often). See https://github.com/StanfordSNR/puffer/blob/master/src/opus-e... if you are interested for more.
I guess a complementary question is in order: since you believe that the WebRTC protocol itself is not incompatible with the mechanisms that enable Salsify's performance, how much work would it be to adapt Salsify incrementally into a WebRTC implementation? (Ignoring mandatory codec support for a moment).
I've been developing an Opus codec mode for Bluetooth A2DP, and I have a similar sort of situation. I've been looking (casually) in to using more interesting rate control based on channel performance to improve QoS with Bluetooth A2DP. I think Opus has a property you could only dream of: strict limits on frame size. :- )
Added: just found that the video codec uses the VP8 bitstream, that was not immediately clear to me; seems to me that that is a substantial selling point that should be made more clear on the webpage.
Since your interface doesn't require any change to the bitstream of a conventional codec, it's a heck of a lot closer to public use than I initially thought!