Need to impliment WebRTC support

Is it possible to make the Asterisk to utilise an existing SIP only framework and provide the WebRTC support. We already have an existing SIP only framework, which handles the media streaming. Hence we need to impliment the WebRTC support over that existing platform by providing the supports for ICE, DTLS, STUN and TURN

I don’t quite know what you mean by “existing SIP only framework”. If you mean use the media layer in Asterisk, then you’d need to still speak SIP or some other protocol. As well if you’re doing pure SIP and don’t need the application layer that Asterisk supports then something like Kamailio with rtpproxy would likely be a better fit as it would purely do a conversion with less overhead.