Thanks — I’m responding to your question about ensuring a short hold message plays fully before transferring the caller to a human agent. Short answer • Play the message with

session.say()

or

session.generate_reply()

and then await the returned SpeechHandle’s playout completion (use

wait_for_playout()

/

waitForPlayout()

) before calling the SIP transfer. That guarantees the audio finishes before the call is connected to the agent. Why this works •

session.say()

/

session.generate_reply()

return a SpeechHandle. Calling

await handle.wait_for_playout()

ensures the spoken audio is delivered completely to the participant. If you don’t await it, the transfer will run immediately and cut off playback. Examples 1) Python

handle = session.generate_reply(instructions="Please hold while I connect you to an agent.")
await handle.wait_for_playout()  # wait until playback finishes

await job_ctx.api.sip.transfer_sip_participant(
    api.TransferSIPParticipantRequest(
        room_name=job_ctx.room.name,
        participant_identity=caller_identity,
        transfer_to=f"tel:{human_agent_number}",
    )
)

2) Node.js / TypeScript

const handle = session.generateReply({ instructions: 'Please hold while I connect you to an agent.' });

await handle.waitForPlayout(); // wait until playback finishes

await jobCtx.room.sipTransferParticipant({ transferTo: `tel:${humanAgentNumber}` });

Additional notes • You do not need to change interruption settings just to ensure playout completes — sequencing the play + await + transfer is sufficient. • If you want to support cancelling the transfer while the message plays (e.g., agent becomes available earlier), you can keep a reference to the SpeechHandle and cancel or check its state before transfer. If you’d like, tell me which SDK (Python, Node, or another) and I can provide a copy-paste-ready snippet integrated into your transfer flow. Sources: Agent speech and audio | LiveKit Documentation | How to detect when an agent has finished speaking | LiveKit