Realtime Tokens — Removed

Removed in MVP

The legacy POST /v1/realtime/token and POST /v1/webrtc/token endpoints have been removed. They now respond with 410 Gone and a migration payload pointing at the canonical Calls API.

{
  "error": "Gone",
  "code": "realtime_token_removed",
  "message": "POST /v1/realtime/token has been removed. Initiate WebRTC sessions via POST /v1/calls with a webrtc participant.",
  "migration_url": "https://docs.rymi.io/api/calls#webrtc"
}

Why the change

/v1/realtime/token minted LiveKit credentials and inserted bare calls rows directly, bypassing the credit, quota, admin-pause, and publishable-key channel checks enforced by POST /v1/calls. Routing browser/SDK voice sessions through /v1/calls keeps every call subject to the same gates.

How to migrate

Send a single request to POST /v1/calls with a webrtc participant. The response includes the same LiveKit credentials the legacy route returned, alongside the canonical call_id and any participant rows you'd otherwise have to backfill.

curl -X POST https://api.rymi.live/v1/calls \
  -H "Authorization: Bearer YOUR_API_KEY" \
  -H "Content-Type: application/json" \
  -d '{
    "agent_id": "550e8400-e29b-41d4-a716-446655440000",
    "participants": [
      { "transport": "webrtc", "identity": "browser" }
    ],
    "metadata": { "source": "web-app" }
  }'

The participants[0].access block in the response contains the LiveKit credentials:

{
  "id": "9b3a4b7c-...",
  "room_name": "call_9b3a4b7c-...",
  "status": "queued",
  "participants": [
    {
      "id": "...",
      "transport": "webrtc",
      "identity": "browser",
      "status": "joining",
      "access": {
        "url": "wss://livekit.rymi.live",
        "token": "eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9..."
      }
    }
  ]
}

Use access.url and access.token exactly where you previously used server_url and client_token.

import { Rymi } from '@rymi/web';

const call = Rymi.createCall({
  url: response.participants[0].access.url,
  token: response.participants[0].access.token,
});
call.on('started', () => console.log('Connected'));
await call.connect();

See the full Calls API reference for the request/response schema, including credit gating, publishable-key channel scoping, and admin-pause behaviour.

Turn-taking and speech boundary detection

WebRTC sessions go through the same hosted agent runtime as PSTN sessions, so the agent's configured turn detector applies to both. Pick a mode by Rymi-native name — steady, balanced, sensitive, or provider — using the agent's advanced turnDetectorMode field. Engine labels (rms, webrtc, silero, provider-signal) are accepted as compatibility aliases and are not the preferred public language.

Mode	Engine on hosted runtime	When to use
steady	RMS, no native dependency	Default. Conservative, no extra installs.
balanced	Native WebRTC VAD when available, PCM fallback	General-purpose latency/false-alarm tradeoff.
sensitive	ONNX Silero when available, PCM fallback	Catches brief utterances and partials more aggressively.
provider	Provider/recognizer speech-boundary signals	STT emits the turn boundary directly.