Thanks — I’m sorry you hit that 400 error while trying to transfer a call. I’ll help you narrow down the cause and get this working. You reported this error: {"message":"Transfer failed: ('invalid_argument','twirp error unknown: sip status 400: BAD_REQUEST')"} A SIP 400 BAD_REQUEST during transfer usually means the transfer request was malformed or contains invalid parameters. To narrow this down, please check/provide the items below; I’ve included targeted checks and quick fixes you can try right away. 1) Confirm the transfer destination format • The

transfer_to

must be a proper SIP URI (sip:user@host) or a

tel:

URI with the correct international format. • Examples that work: ◦

<sip:+15551234567@your-sip-host.com>

◦

tel:+15551234567

• Common problems: ◦ Missing schema (e.g.,

+15551234567@host

instead of

sip:+...

) ◦ Missing

+

for E.164 numbers (e.g.,

tel:15551234567

) Action: Share the exact

transfer_to

value you’re using (mask sensitive parts if needed) or try replacing it with a simple

tel:+15551234567

test value. 2) Verify participant identity and room name • The

participant_identity

and

room_name

in the transfer request must exactly match the SIP participant currently in the room. Action: Confirm the values you send in

TransferSIPParticipantRequest

and, if possible, run

list_participants

for the room to confirm the participant identity and that the room is active. 3) Ensure your SIP trunk supports transfers (SIP REFER) • Transfers require that your SIP trunk/provider supports SIP REFER (and that transfer is enabled). Action: Check your SIP trunk/provider settings (e.g., Twilio: “Call Transfer (SIP REFER)” and “Enable PSTN Transfer” must be enabled) and share whether your provider supports REFER. 4) Check API credentials and request construction • Make sure the LiveKit API client used for transfer is initialized with valid credentials and you’re constructing the request according to your SDK. Action: Confirm the API client is initialized with the correct

LIVEKIT_URL

,

LIVEKIT_API_KEY

, and

LIVEKIT_API_SECRET

, and log the full transfer request payload just before sending it. 5) Quick debugging steps to run now • Log the full request payload (mask secrets) immediately before the transfer call. • Try a minimal test: transfer to a

tel:

URI in E.164 format to rule out SIP URI formatting. • Check the SIP provider/trunk logs at the timestamp of the failed transfer for more detailed SIP error information. If you can, please provide: • The exact (masked)

transfer_to

value you used. • The

participant_identity

and

room_name

from the transfer request. • Which SIP provider/trunk you use and whether REFER transfers are enabled. With those details I’ll pinpoint the likely cause and give the exact fix. If you prefer, paste the logged transfer request here (mask any credentials) and I’ll review it. Sources: Pass UUI data to Genesys Cloud during SIP transfers | LiveKit | SIP Lifecycle Management Agent | LiveKit Documentation | Company directory phone assistant | LiveKit Documentation | Agents telephony integration | LiveKit Documentation