Glossary
WebRTC
Also called: Web Real-Time Communication
WebRTC (Web Real-Time Communication) is an open standard, built into modern browsers and available to native apps, for sending audio, video and data between devices in real time without plugins.
Reviewed
What WebRTC includes
WebRTC is two sets of standards working together: browser APIs from the W3C and network protocols from the IETF. In practice it gives you:
- Capture: access to the camera and microphone (getUserMedia).
- Connections: RTCPeerConnection, which handles codecs, bandwidth adaptation and network traversal through ICE, STUN and TURN servers.
- Data channels for sending arbitrary data — chat, files, game state — over the same connection.
- Mandatory encryption: media is encrypted with DTLS-SRTP. There is no unencrypted mode.
How a WebRTC call is set up
- 1
Signaling
The caller creates an offer describing its media (SDP) and sends it through the app's own server — often a WebSocket. WebRTC deliberately doesn't define this part.
- 2
Answer
The other device replies with an answer. Both now agree on codecs and formats.
- 3
ICE
Both sides gather candidate addresses and test pairs, preferring a direct peer-to-peer route and falling back to a TURN relay.
- 4
Media flows
A DTLS handshake sets up encryption keys, and encrypted audio and video start flowing.
Group calls: mesh, SFU and MCU
| Topology | How it works | Trade-off |
|---|---|---|
| Mesh | Every participant sends their stream to every other participant | No media server, but upload grows with each person; suits small calls |
| SFU | Each person sends one stream to a server that forwards it to the others | Scales well; the server handles media |
| MCU | A server mixes all streams into one | Easy on devices, heavy on servers; adds delay |
- Mesh
- How it works:Every participant sends their stream to every other participant
- Trade-off:No media server, but upload grows with each person; suits small calls
- SFU
- How it works:Each person sends one stream to a server that forwards it to the others
- Trade-off:Scales well; the server handles media
- MCU
- How it works:A server mixes all streams into one
- Trade-off:Easy on devices, heavy on servers; adds delay
On Asyncbit
Asyncbit's voice and video calls are built on WebRTC, connecting directly where the network allows and relaying through TURN when it doesn't, so calls connect on mobile data and restrictive networks. You start a call from a chat, 1:1 or group, in the browser or the Android app (which supports picture-in-picture). Group calls hold up to 8 people, with video for up to 4; larger calls continue as audio. Call history and a summary line in the chat record each call.
See it for yourself
Asyncbit is free, with no paid tier and no ads. Everything described here is available from the first day.
Create your free accountRelated terms
Related Asyncbit features
Where this comes up
- Use caseGroup chat & callsGroup chats for 3–50 people with voice messages, reactions and calls for up to 8 (video for 4). How it works on Asyncbit, and when another app fits better.
- GuideHow to make video calls work on a bad networkWhy video calls freeze, drop or never connect, how NAT, STUN and TURN relays decide whether a call gets through, and fixes that work in any calling app.