webrtc

Softswitch with WebRTC support:

    asterisk (for linux)
    freeswitch (linux and windows)
    mizuvoip (for windows)
    kamailio (for linux)

WebRTC to SIP gateways:

    doubango
    oversip
    mrtc
    janus
    oracle

SIP (RFC 7118) capable WebRTC clients:

    Simpl5 (from dubango)
    Jssip (old good)
    Sip.js (successor of Jssip)
    Webphone (works also in IE and Edge)
    jscommunicator (with drupal plugin)

Also you should deploy and use your own STUN and TURN servers (some of the server/gateways have these built-in, otherwise use coturn rfc5766-turn-server).

Once the server side is up and running, you can easily create your custom client side solution based on the above webrtc clients since each of them has a simple to use JavaScript API.