[SR-Users] SIP - WebRTC gateway

Hello, I'm figuring out the best approach to deploy a bridge between Websocket\Webrtc and SIP\rtp. Can Kamailio (+mediaproxy-ng or something else) operate as a full Webrtc\SIP gateway (signaling, audio or video transcoding, ICE and so on)? Some months ago I found the architecture described here http://www.kamailio.org/wiki/devel/rtcweb_breaker that proposes to introduce a new RTCWeb Breaker. Is it just a proposal or is Kamailio moving following this approach? If Kamailio really requires a RTCWeb Breaker, what are the main issues against using Doubango webrtc2sip with Kamailio? Performance? Interoperability? License? ... Thanks Daniele ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. **********************************************************************