For LiveKitAI Platform

Telephony Sip

LiveKit (Cloud + Agents) · LiveKit

Real-time Voice & Video Infra — LiveKit

LiveKit evals — Telephony (SIP) (relift v3 InfraRed)

About LiveKit

LiveKit is open-source real-time voice/video infrastructure used to build voice agents and live experiences — a WebRTC SFU, telephony (SIP), recording/egress, and the LiveKit Agents framework for STT→LLM→TTS pipelines, available as LiveKit Cloud and self-hosted.

Employees

~50

Industry

Voice AI Infrastructure

Headquarters

New York, NY

Website

livekit.io

Sample tests· showing 3 of 9

#	Input	Expected behavior	Check
01	Two dispatch rules match an inbound number — Rule A (direct to room 'support') and Rule B (individual rooms per call).	Per LiveKit SIP docs, dispatch rules are evaluated in order; the first match wins. Inspect the configured ordering and make precedence explicit. Do not rely on creation-time order across deployments; specify it in the IaC.	Pass / FailAi Platformhigh
02	Server initiates an outbound call: CreateSIPParticipant({trunk_id, sip_call_to:'+15551234567', room_name:'call-xyz', participant_identity:'callee'}).	Verify the trunk is configured for outbound (CreateSIPOutboundTrunk done); the trunk auth (username, password, address) matches the carrier; from-number is an authorized caller ID with the carrier. Surface the carrier's SIP response (486 busy, 480 unavailable, 603 declined) back to the operator — d…	Pass / FailAi Platformcritical
03	SIP call has packet loss on the carrier leg only (LiveKit ↔ SFU is healthy).	LiveKit SIP service bridges RTP between the carrier and the room; loss on the carrier leg shows up as RTP loss on the SIP participant's track. Use the SIP participant's stats (jitter, packet_loss) to isolate carrier-side issues from room-side issues. Escalate to the trunk provider with their call S…	Pass / FailAi Platformmedium
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