Low cost, monthly, per user solution enabling SIP signaling for RTP Voice transactions.
With Internet Explorer being decommissioned, support for ActiveX is rapidly in decline. A complete phase-out is on the very near horizon.
For organisations offering voice solutions, setting up new WebRTC servers is capex heavy, expensive to maintain, and can interfere with key services like call recording and PCI compliance solutions.
iNet provides WebRTC functionality directly to communications providers as-a-service; and at a simple, low per-user monthly charge. Carrier-agnostic, clustered and resilient infrastructure design, global connectivity, zero latency, and NO headache.
WebRTC controls the signalling connection between two peers: The user, and the PBX, for example. Minimal bandwidth is required to setup this signalling as the voice audio path and RTP compression is only established between the PBX and the user connection and sits outside of the WebRTC connectivity path.
For an inbound call, the PBX sends a SIP invite to our WebRTC server. For an outbound call, the agent/user initiates the SIP invite. The WebRTC server then initiates a SIP session with the agent/user. Once both signaling paths are established, the RTP call is connected between the PBX and agent/user, outside of the WebRTC signaling path.
