WEBRTC-SIP GATEWAY

The WebRTC-SIP (ProlopeRTC) gateway will make your IP-PBX or softswitch WebRTC capable, allowing desktop and mobile browsers to initiate and receive calls to/from your SIP service over websocket and WebRTC completely transparently, without any configuration changes on your existing server(s). On the server side it is compatible with all PBX/VoIP server/SIP trunk/proxy/gateway/carrier supporting the SIP protocol such as Asterisk, 3CX, Broadsoft, Brekeke, Yate, FreePBX, Elastix, Trixbox, Voipswitch, FreeSWITCH, Cisco, Siemens, Huawei, NEC, Mitel and others. Multiple SIP server support (route calls to one or more SIP server, accept calls from one or more sip servers). Direct SIP peers are also supported. On the client side you can use any library implementing WebRTC and SIP over WebSocket as specified in RFC 7118, compatible with WebRTC stacks present in browsers like Chrome, Firefox, Edge, Opera and others, WebRTC plugins for IE or Safari or native libraries such as PJSIP. All the common WebRTC SIP clients and JavaScript WebRTC libraries are supported such SIPML5, JSSIP, SIP.JS and others. Works from smartphone, tablet or desktop, using any operating system (Windows, Linux, MAC, Android, iOS). Beside browser to SIP and SIP to browser, the gateway also has full support for browser to browser calls and SIP to SIP calls.