websocket – Difference beween Sip with webrtc and Sip trunking providers in voip calling

I just started learning sip and i am very confused between Sip with Webrtc and Sip trunking providers.

So far i have understood that sip providers sit between two peers for providing communication channel and this means the connection isn’t peer to peer.

I have used Webrtc before and i was also able to do sip over Websockets with Webrtc. I understand how Webrtc is peer to peer.

But i also read some resources that state that Webrtc and Sip trunking can be used in conjunction. But aren’t both of them used for Media streaming, so how would we go about using them together besides Sip trunking is digital analog of legacy phone lines so what does it have to do with webrtc.

Also is sip trunking only used for legacy telephone system? If not how can i develop a voip calling app with sip trunking in javascript i did not find any resources related to this.