Latency versus reliability is a false dichotomy anyway. The alternative to WebRTC isn't to wait for the user to finish speaking before you send any of the audio. Open a websocket and send the coded audio packets as they're generated. Now you're still sending audio packets immediately, but if one is dropped, TCP retransmits it until it makes it through. If the connection is really slow, packets queue up, and the user has to wait, but it still works. You get the low latency in the best case and the robustness in the worst case.