Calls

A call needs two things: REST to place, answer and hang up, and a WebRTC session that carries the sound and reports what happens to the call on its data channel.

1. Open a WebRTC sessionCopy Anchor Link

Create a peer connection with the ice_servers returned with your ephemeral key, add one audio track and a data channel labelled 06line, gather every ICE candidate, then send the offer:

const pc = new RTCPeerConnection({ iceServers });
pc.addTrack(microphoneTrack);
const events = pc.createDataChannel('06line');

await pc.setLocalDescription(await pc.createOffer());
await new Promise((done) => (pc.onicegatheringstatechange = () => pc.iceGatheringState === 'complete' && done()));

const res = await fetch('https://api.06line.com/v1/webrtc/sessions', {
  method: 'POST',
  headers: { Authorization: `Bearer ${ephemeralKey}`, 'Content-Type': 'application/json' },
  body: JSON.stringify({ sdp: pc.localDescription.sdp }),
});
const session = await res.json(); // { id: "rtc_…", sdp }
await pc.setRemoteDescription({ type: 'answer', sdp: session.sdp });

There is no trickle ICE: the offer must hold every candidate. Audio is G.711 (PCMU or PCMA), 8 kHz, which every browser supports; keep the offer's default codecs and 06 Line picks it.

Keep the session open as long as the user may call or be called: it carries the audio of one call at a time, call after call. Close it with DELETE /v1/webrtc/sessions/{id} or by closing the peer connection.

2. Listen to the data channelCopy Anchor Link

06 Line sends JSON messages on the 06line channel:

{ "type": "call.state", "data": { "id": "call_…", "state": "active", "answered_at": "2026-10-21T15:12:11Z" } }

Every open session of a company receives call.incoming: ring everywhere, and the first session to answer takes the call. The channel is for receiving only; actions go through REST.

3. Place a callCopy Anchor Link

curl -X POST https://api.06line.com/v1/calls \
  -H "Authorization: Bearer $KEY" \
  -H "Content-Type: application/json" \
  -d '{"line_id": "line_…", "to": "+33698765432", "session_id": "rtc_…"}'

The call starts in dialing; call.state then reports ringing, active and ended.

4. Answer a callCopy Anchor Link

On call.incoming, answer with the session that should carry the sound:

curl -X POST https://api.06line.com/v1/calls/call_…/answer \
  -H "Authorization: Bearer $KEY" -H "Content-Type: application/json" \
  -d '{"session_id": "rtc_…"}'

Without an answer, the call ends with end_reason: "no_answer".

5. Hang upCopy Anchor Link

POST /v1/calls/{id}/hangup ends the call, or declines it while it rings.

One call at a time per line and per session: a second one fails with line_busy.

RecordingsCopy Anchor Link

Every call is recorded, from the moment it is placed or answered until it ends, as two mono WAV files (PCM 16-bit, 8 kHz), one per speaker:

• GET /v1/calls/{id}/recordings/remote: the other party, at the end of the phone line;
• GET /v1/calls/{id}/recordings/local: what your WebRTC session sent.

Both start at the same moment, so they line up: play them together, or transcribe each speaker on its own. They are ready a few seconds after the call ends.

Rather than polling, subscribe a webhook endpoint to call.recorded: it arrives once the files are stored, with the call and a presigned link to each track, valid 7 days and usable without your API key, ready to hand to a transcription service.

Native appsCopy Anchor Link

Nothing here is specific to browsers: the same session and REST calls work from any WebRTC stack.

The data channel only lives while your app runs. Ringing a phone whose app is closed needs a push notification (PushKit and CallKit on iOS, FCM and ConnectionService on Android); 06 Line does not send those yet.

HistoryCopy Anchor Link

GET /v1/calls lists past and current calls, newest first, filtered by line_id and/or peer (the other party's number), exactly like SMS history.