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createLiveKitSipCallProvider controls a SIP participant using borrowed LiveKit server SDK ports. It is separate from LiveKitVoiceTransport, which handles room audio.

Prerequisites

Provision an outbound SIP trunk and a room, then create authenticated server clients in your host. Pass structural LiveKitSipCallPort and LiveKitCallRoomPort implementations to Agentium. No LiveKit SDK or credentials are loaded by the adapter itself.

Bind the route

Use one host-controlled route per room. Do not derive the room or trunk from a caller’s dial payload. Add this provider to OutboundCallService, which enforces intent ownership and authorization around every operation.

What the adapter owns

A create request assigns a deterministic call-… participant identity and sets waitUntilAnswered: false and hidePhoneNumber: true. The call reference includes the room, participant identity, and SIP call ID. Reads verify the participant’s SIP call identity before reporting status. Hangup removes that SIP participant. It does not close the room or your other participants. A missing participant does not automatically become a completed call; the host needs reliable terminal evidence. Your application owns trunk provisioning, room connection, track publication, audio conversion, callback verification, and client lifecycle. See LiveKit outbound calls and media transport for those separate connections.