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Is there a way to make an outbound dial from a liv...
# ask-ai
w
Is there a way to make an outbound dial from a livekit room to the pstn without an elastic sip trunk, and instead using a programmable voice app? I want to be able to use create_sip_participant in my code, but I also need to be able to use the caller id of an inbound caller using calltoken in twilio on that dial
t
Thanks for reaching out, Drew. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
Thanks — I’m responding to your question about making an outbound PSTN dial from a LiveKit room using a programmable voice app (Twilio) instead of an Elastic SIP Trunk, and whether you can use 
create_sip_participant
while preserving the inbound caller’s ID (from a Twilio call token). Short answer • Yes. You can place outbound PSTN calls from a LiveKit room using 
create_sip_participant
and route the outbound leg through Twilio’s SIP trunking (i.e., Twilio as your SIP provider) rather than requiring an “Elastic SIP Trunk.” • You can pass the inbound caller’s number as the outbound caller ID by setting the 
sip_number
(caller ID/ANI) in the 
CreateSIPParticipantRequest
, provided Twilio accepts that caller ID for your trunk configuration and the number meets Twilio’s policies. How this typically works 1. Capture the inbound caller’s number from the incoming SIP leg (the 
From:
or ANI on the inbound call). 2. Call LiveKit’s 
create_sip_participant
for the room, specifying: ◦ the Twilio-connected SIP trunk (configured in LiveKit), ◦ the destination PSTN number to dial (
sip_call_to
or equivalent), ◦ and 
sip_number
set to the original inbound caller’s number so the outbound leg presents that caller ID. 1. Twilio’s trunk receives the SIP call from LiveKit and routes to the PSTN. Twilio will honor the 
From:
/ ANI if your trunk settings allow caller-ID pass-through and the number is permitted by Twilio. Important constraints and setup notes • Twilio authentication: Twilio trunks typically expect SIP credentials or IP ACL-based authentication; LiveKit needs to be configured to authenticate to the Twilio trunk. Twilio call tokens (used in some Twilio SDK flows) are not directly part of SIP signaling—SIP routing uses trunk auth and headers. • Caller ID acceptance: Twilio must allow caller-ID passthrough for your trunk. You may need to enable caller ID transfer/passthrough in your Twilio trunk settings and ensure the number you want to present is permitted/verified under Twilio rules. • SIP headers to set: Use 
From:
and/or 
P-Asserted-Identity
as appropriate. Setting 
sip_number
in the 
CreateSIPParticipantRequest
instructs LiveKit to populate the outbound SIP caller ID. • Compliance: Twilio may block arbitrary spoofing; the number used as caller ID may need verification or specific trunk/policy settings. Next steps I can help with • If you want, share which fields you currently see in your 
create_sip_participant
request (room name, trunk ID, fields you can set). I can give a concrete example request for your code language. • If you’re using Twilio trunks and want the exact Twilio settings to enable caller-ID passthrough, tell me whether your trunk uses SIP credential auth or IP ACLs and I’ll outline the settings to check. • If you prefer, I can draft a sample 
create_sip_participant
call showing where to put 
sip_number
and the dial target. Would you like a code example in Python or JavaScript for 
create_sip_participant
with 
sip_number
filled in? Sources: SIP Lifecycle Management Agent | LiveKit Documentation | Outbound Calling Script | LiveKit Documentation | Agents telephony integration | LiveKit Documentation | Preserving Caller ID During SIP Transfers in LiveKit Agents | Preserve caller ID during SIP transfers with Twilio | LiveKit
w
hm, is there a way to use create_sip_participant (or something similar) without an elastic sip trunk, and just go straight to a programmable voice app?
t
Thanks — you’re asking whether you can use 
create_sip_participant
(or something similar) to dial outbound to the PSTN from a LiveKit room without any SIP trunk at all, and instead go straight to a programmable voice app (e.g., Twilio Programmable Voice). Short answer • No — you cannot bypass a SIP endpoint/trunk entirely when using LiveKit’s SIP participant functionality. LiveKit’s outbound PSTN dialing uses SIP signaling, so you need a SIP target for the outbound leg. That means either a SIP trunk (Twilio SIP trunk, a carrier SIP provider, or your own SIP gateway) or a SIP-capable endpoint that the programmable voice app can accept. • You can, however, use Twilio Programmable Voice in combination with SIP (Twilio acting as the SIP target) rather than an “Elastic SIP Trunk” product specifically — Twilio can act as a SIP provider/trunk that LiveKit dials into. This effectively uses Twilio (programmable voice) for PSTN routing while still sending/receiving SIP from LiveKit. Why a SIP trunk/endpoint is required • LiveKit creates an outbound SIP leg when you call 
create_sip_participant
. That API expects to send SIP to some SIP destination. Programmable voice apps like Twilio accept and route calls via SIP trunks or SIP interfaces — LiveKit needs to send SIP to Twilio’s SIP interface to reach Twilio’s PSTN routing. • Twilio’s client-side call tokens (e.g., a Twilio Call Token used in Twilio SDKs) are not a SIP target by themselves; they’re credentials for Twilio SDK flows. SIP signaling is the protocol LiveKit speaks for telephony participants. Practical options • Use Twilio as the SIP target/trunk: ◦ Configure a Twilio SIP trunk or SIP domain to accept calls from LiveKit (via IP ACL or SIP credentials). ◦ Call 
create_sip_participant
in LiveKit pointing at that Twilio SIP trunk and set 
sip_number
to present caller ID as needed. ◦ This uses Twilio Programmable Voice for PSTN without needing a separate “Elastic SIP Trunk” product. • Use another SIP gateway or provider: ◦ Point LiveKit at any SIP-capable gateway/service that will route to the PSTN. • Run your own SIP gateway: ◦ If you want more control, run a lightweight SIP gateway that accepts LiveKit’s SIP and then uses the programmable voice app’s REST APIs or SDKs to place outbound PSTN calls. Note: this reintroduces a SIP-to-API bridge you must operate. • Peer with a Twilio SIP interface that accepts SIP INVITE and then Twilio handles the PSTN leg — this is functionally equivalent to using Twilio as your SIP trunk. If you want to avoid any SIP at all (no SIP trunk, no SIP gateway), LiveKit doesn’t provide a native direct-to-Twilio-Programmable-Voice REST flow for outbound PSTN. You would need a bridging component that translates LiveKit’s SIP into the programmable voice app’s API calls. Would you like: • A concrete example showing the 
create_sip_participant
fields you’d set to point at a Twilio SIP trunk? • Guidance on configuring Twilio (SIP domain/trunk) to accept LiveKit calls (IP ACL vs credentials, caller ID passthrough)? • A sample architecture for a lightweight SIP->Twilio-API gateway if you prefer not to use Twilio’s SIP trunking?
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