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Supported platforms and browsers
Chrome | Firefox | Safari 11 | Edge | |
|---|---|---|---|---|
Windows | + | + | + | |
Mac OS | + | + | + | |
Android | + | + | ||
iOS | - | - | + |
Operation flowchart
The browser connects to the server via the Websocket protocol and sends the publish command.
The browser captures the microphone and the camera and sends a WebRTC stream to the server over TCP.
- The second browser establishes a connection also via Websocket and sends the play command.
- The second browser receives the WebRTC stream over TCP and plays that stream on the page.
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By default, buffers sizes are set to 1 M.
Adjusting TCP queues
TCP queues high and low watermarks are set with the following parameters
| Code Block | ||
|---|---|---|
| ||
ice_tcp_channel_high_water_mark=52428800
ice_tcp_channel_low_water_mark=5242880 |
By default, TCP queues size is between 5242880 and 52428800 bytes.
Ports used
TCP ports from WebRTC media ports data range are used for WebRTC connection over TCP
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