Proxy traffic from classic SIP Softphone to WebRTC/websocket

I’m pretty new to the VOIP world, so I apologize if this has been discussed elsewhere. I used both forum and google search and was unable to locate information about my topic.

I am involved in a project where we would like to proxy or somehow route/gateway traffic from a number of classic SIP phones (hardware and softphone) to a WebRTC/SIP-over-websocket server that our provider has setup for us. Can Asterisk help us out?

Here’s a rough ASCII diagram of what we want to do:
SoftPhone => Asterisk => WebRTC/SIP-over-websocket-server

The softphones would rely on Asterisk to know how to speak to to the providers server and generally act as a communications relay.

Any examples, tutorials or advice is welcome. :smile:

Asterisk cannot proxy.

If the right hand side is within the scope of the parts of WebRTC that Asterisk supports, using Asterisk as a back to back user agent should be trivial, requiring only Asterisk 101 level knowledge.

This forum is a discussion forum; it is not for asking questions.