Hi! I want to use the same SIP friend account and to be able to connect to this account from WebRTC and from ordinary sipphone. The problem is that WebRTC requires some sip options (like encryption=yes and DTLS), which are not comparable with my sipphones. Am I able to say to asterisk to pickup webrtc-specific options only if user is connected/registered via ws/wss transport?
Of course I can use some hacky solutions like webrtc2sip, but I think this is not good.
You’re going to be better off, methinks, not trying to do that. Rather, you’ll gain more flexibility that’s specific to the endpoint by utilizing separate friends/peers for each different device you’ll be bringing in.