{
  "id": "channel:sip-telephony",
  "_kind": "Channel",
  "_file": "channels-hooks/channels/voice-transport.yaml",
  "_cluster": "channels-hooks",
  "attributes": {
    "displayName": "SIP / PSTN Telephony",
    "kind": "voice",
    "endpoint": "sip",
    "persistent": true,
    "description": "Realtime phone transport (SIP/PSTN, inbound + outbound, DTMF, REFER, warm/cold\ntransfer) bridged to WebRTC by the external SIP gateway (livekit/sip). Inbound\ncalls map onto the channels-adapter event->bounded-session-spawn pattern. See\ndocs/research/realtime-voice-agent-stack.md.\n"
  },
  "outgoingEdges": [
    {
      "from": "channel:sip-telephony",
      "to": "layer:3-transport",
      "kind": "realizes",
      "attributes": {}
    }
  ],
  "incomingEdges": []
}