I'd hope so, considering they were a part of the group standardizing it. Opus, in broad strokes, is Skype's SILK codec for low bitrates, Xiph's CELT for high bitrates, and a mode that is a blend of both codecs for the middle.

It might. Using Opus does not equate to low audio latency. (I don't actually know Opus, but I have passing familiarity with it's predecessor speex). Assuming Opus has the same design goals as speex, Opus gives you the tools necessary to implement a low-latency audio application. As in - the audio codec piece is "solved", now you need to do the engineering work around client/server/framing/network etc. to actually get to your latency goals.