Questions regarding WebRTC and configurability of Asterisk scripts

We’re a software company with a webapp that has a softphone integrated in it using Websockets and WebRTC and an API system that IP PBXs can connect with a custom dial plan solution for robot answering.
We have a few questions, specifically for Grandstream UCM6301 (we’ve never connected to a Grandstream PBX before).
Firstly, is it possible to connect to it via WebRTC, what would the websocket URL be?
And secondly, is there any way to configure the PBX inbound routes to go to a custom Asterisk dial plan which uses custom AGI script files with Perl?

Thanks in advance!