WebRTC — Web Real-Time Communication
What is WebRTC?
WebRTC (Web Real-Time Communication) is the open, royalty-free browser standard for streaming live audio and video, or a data channel, directly between a browser or mobile app and a server with no plugin or native app install required. It handles media capture, network traversal (via STUN/TURN and ICE), and encryption (SRTP) under a single JavaScript API, which is what lets a browser-based softphone place a call or a video room stream live media the moment a page loads.
More detail
WebRTC still relies on SIP-style session negotiation and RTP for the actual audio/video stream underneath — it standardizes the browser-side capture, transport, and encryption layer, not a replacement for those underlying voice protocols.
Because a browser can't always reach another endpoint directly (NAT, firewalls), WebRTC uses STUN to discover a usable public address and falls back to relaying media through a TURN server when a direct path isn't possible.
Orbit's browser softphone and in-browser video rooms run on WebRTC, so a call or video session starts from a join link or the dashboard with nothing for the user to install.
Frequently asked
- Do I need to install anything to make a WebRTC call?
- No — that's the point of WebRTC. Audio and video stream directly from the browser or app using a built-in API, with no plugin or separate native application required to join a call or video room.
- Is WebRTC the same thing as SIP?
- No. SIP is a signaling protocol for setting up and tearing down a call; WebRTC is a browser standard that handles media capture, encryption, and network traversal for real-time audio, video, and data — a WebRTC call still typically negotiates its session using SIP-style signaling underneath.
See also
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