I know FreeSWITCH and Kamailio both have WebRTC connections available, so if you use a SIP client in the browser over WebRTC to the server, you can plug into telephony networks on the server side.
We use Kamailio's WebRTC implementation heavily in Kazoo along with our libwebphone client. The transport is abstracted so Kazoo deals with the device and its configs; the Kamailio instance the browser connects to does the TLS termination for WebRTC. FreeSWITCH has the smarts for the SDP DTLS bits. And it all just works real nice together.
We use Kamailio's WebRTC implementation heavily in Kazoo along with our libwebphone client. The transport is abstracted so Kazoo deals with the device and its configs; the Kamailio instance the browser connects to does the TLS termination for WebRTC. FreeSWITCH has the smarts for the SDP DTLS bits. And it all just works real nice together.