Build webRTC2sip proxy (Kamailio/Asterisk)

I want to enable sip calls directly from a web browser (Chrome, Firefox, IE) without installing any plug-ins. I am looking for someone to build a WebRTC proxy (with java SIP libraries) to sit between the web browser and my Kamailio SBC (or asterisk). I want to enable voice and video calls as well as instant messaging (browser to browser and browser to PSTN). I would like to have a step by step manual produced how to install and how to configure the proxy (preferably on Debian Wheezy).