This is using sipgo as a back-to-back user agent (B2BUA). There are really two sessions going on: one between the SIP caller and sipgo, and a second between sipgo and the browser. These sessions each have their own signaling and media.

The interesting part is how the signalling is happening between the sipgo server and the browser. You copy your session description from the browser and paste it to the server! In a "real" application, you could send it over a custom API call or something, or you could use a standard protocol for initiating sessions (SIP!).

This is all to say, "bridging" between SIP and WebRTC is not nearly as magical as it is often made out to be. In some cases, the media can even be passed through without modification. (Although many SIP endpoints do not support e2ee and so a B2BUA is still needed to wrap/unwrap the encryption.) We were doing this a decade ago with SIP.js and OverSIP.