WebRTC is best for web applications, mobile apps, and end-user facing interfaces with ultra-low latency and built-in NAT traversal.Documentation Index
Fetch the complete documentation index at: https://docs.deepslate.eu/llms.txt
Use this file to discover all available pages before exploring further.
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.
| Format | Sample Rate | Channels | Notes |
|---|---|---|---|
| Opus | 48000 Hz | Mono | Browser default |
| Opus | 16000 Hz | Mono | Mobile optimized |
Error Handling
Connection Errors
Connection Errors
| Code | Description | Resolution |
|---|---|---|
auth_failed | Invalid or expired API key | Check your API key |
rate_limited | Too many concurrent sessions | Reduce connection rate |
invalid_config | Invalid assistant/agent ID | Verify resource exists |
Runtime Errors
Runtime Errors
| Code | Description | Resolution |
|---|---|---|
ice_failed | ICE connection failed | Check network/firewall settings |
session_timeout | Session idle too long | Reconnect |
internal_error | Server-side error | Retry with backoff |
Browser Compatibility
| Browser | Support |
|---|---|
| Chrome | Full support |
| Firefox | Full support |
| Safari | Full support (iOS 14.3+) |
| Edge | Full support |