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This API is not yet available and is still under construction.
WebRTC is best for web applications, mobile apps, and end-user facing interfaces with ultra-low latency and built-in NAT traversal.

Session Flow

Step 1: Create Session

Request a WebRTC session from the API:

Step 2: Establish Connection

Audio Configuration

WebRTC uses Opus codec at 48kHz by default. Deepslate automatically handles sample rate conversion.

Error Handling

Browser Compatibility