SIP call using WebRTC

I am currently working on configuring webRTC in my local asterisk server (version 13). I have tried to test my configuration using demo client apps like JsSIP and SIPml5. But they required secured connection (https). So I decided to develop my own client application using core WebRTC apis without using any SIP libraries. Is it possible to develop a webRTC client application without using any SIP libraries?

Asterisk does not provide an alternative for signalling for WebRTC. It would be up to you to add such functionality to Asterisk.

@jcolp
Thanks for the response.

Actually I am going to use SIP as the signalling protocol for webrtc configuration. But I don’t want to use JsSIP or SIPml5 libraries . Is this possible. If it is , then please suggest me some ideas.

Thanks.

Asterisk doesn’t care as long as it is SIP. What happens on the client side doesn’t matter.

@jcolp
Thanks for the response.