Join call audio yourself

Instead of the gateway connecting to your agent (streams), your code can connect to the gateway and exchange the call’s audio directly. Useful for scripts, testing, and agents that can’t accept incoming connections.

Connect

wss://<gateway>/v1/calls/{callId}/media
Authorization: Bearer <api key>        (or ?apiKey=<key> where headers aren't possible)

You can connect before the call is answered. Audio starts when the phone starts streaming:

{"type":"start","callId":"8c1d…","format":{"encoding":"pcm_s16le","sampleRate":16000,"channels":1,"frameMs":20}}

Audio

• From the gateway: binary frames of the other person’s voice, 16 kHz mono 16-bit little-endian PCM, 20 ms (640 bytes) each.
• To the gateway: binary PCM in the same format. Any length works; it’s played in order.
• Interrupt: send {"type":"clear"} to drop audio that hasn’t played yet.
• End: {"type":"stop","callId":"…","reason":"device_media_closed"} when the phone stops streaming. Hang up with POST /v1/calls/{id}/hangup.

Only one talking client per call; a new one replaces the old.

import asyncio, json, websockets

async def main(call_id):
    url = f"wss://your-openline/v1/calls/{call_id}/media"
    async with websockets.connect(url, additional_headers={"Authorization": "Bearer gw_..."}) as ws:
        async for msg in ws:
            if isinstance(msg, bytes):
                await ws.send(msg)            # echo the caller back to them
            else:
                print(json.loads(msg))

asyncio.run(main("8c1d..."))

Listen only

Add ?mode=listen to hear both sides without being able to talk (as many listeners as you like). Each binary frame starts with one byte saying whose audio it is:

First byte	Audio
0x01	The other person on the call
0x02	What your agent or code is sending into the call

The rest of the frame is 16 kHz PCM. Listeners also get start, stop and clear messages. The console’s Listen in button uses this.