shy-kitchen-69852
02/13/2026, 10:55 AMtall-airline-16756
02/13/2026, 10:55 AMtall-airline-16756
02/13/2026, 10:55 AMshy-kitchen-69852
02/13/2026, 10:56 AMtall-airline-16756
02/13/2026, 10:57 AM<sip:+1234567890@zoomcrcs.com>), and provide example code using LiveKit’s SIP API (create_sip_participant) to place outbound calls from your agent.
• If you don’t have Zoom Phone: suggest alternative SIP providers that work with LiveKit and outline how to proceed.
• Provide example code for your chosen platform and describe important settings (SIP credentials, caller ID passthrough, SRTP/DTLS, and SIP REFER for transfers).
Reply with the answers above (or even just the one-sentence goal and platform), and I’ll provide the precise steps and sample code.
Sources: Agents telephony integration | LiveKit Documentation | Making calls overview | LiveKit Documentation | Preserving Caller ID During SIP Transfers in LiveKit Agentsshy-kitchen-69852
02/13/2026, 12:39 PMtall-airline-16756
02/13/2026, 12:39 PM<sip:+1234567890@zoomcrcs.com>) when placing calls to PSTN numbers via Zoom Phone.
• In LiveKit, configure an outbound SIP participant/trunk and use LiveKit’s SIP API (e.g., create_sip_participant) to originate outbound calls from your voice AI agents to Zoom’s SIP endpoint.
• Make sure the Zoom SIP trunk settings meet your needs: caller ID passthrough if you want custom caller IDs, SIP REFER support for transfers, and SRTP/DTLS settings for secure media as required.
If you want, I can:
• Walk you through the exact configuration steps for LiveKit (Cloud or self-hosted) and Zoom Phone.
• Provide example code for placing outbound calls from your chosen platform (Node.js, Python, Go).
• Help with SIP credential/auth setup, media security recommendations, or call recording considerations.
Tell me which of the following apply and I’ll provide tailored steps:
• You have Zoom Phone and a SIP trunk already, or you don’t yet.
• You’re using LiveKit Cloud or self-hosted LiveKit.
• Your implementation language (Node.js, Python, Go, etc.).