Create a Linux WebRTC Server to automatically connect players

The experience our team have in VoIP and WebRTC is Developing and maintaining VoIP/Sip applications in .net/c++. Developed WEBRTC Media Gateway in C++. To Link PBX Vendors like Alcatel, Avaya, Cisco and Traditional SIP Platforms with New Web Media Technology. Call from PBX Based Hard Phone or SIP Soft Phone and receive on Web Browser. Also Vice Versa. Developed SIP proxy in C++. Proxy Supports: External (Trunk Network) Routing. B2bUA ( Back to Back User Agent) ,for Intelligent Routing and Call Drop scenarios. Voice Media Gateway to Record Calls. Stateless / Stateful Communication. IVR (Interactive Voice Response). Tried and Tested on Major Telephony Platforms Including AVAYA , Alcatel , CISCO. Full familiarity with Csta/Tapi/Sip protocols. Developed user friendly framework in C++, to support PnP architecture , helping in Rapid application development using C++/C# for quick VoIP integration. Worked on CSTA Phase 1 Protocol for Alcatel OmniPCX Office PBX. Implemented in C# .Net. Worked on CSTA Phase 2 Protocol for Alcatel OmniPCX Enterprise PBX. Implemented in C# .Net