How to Test LiveKit Agents With TestMu AI

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LiveKit is infrastructure rather than a hosted builder. The agent joins a WebRTC room as a participant, and phone calls arrive in that same room through a SIP trunk. TestMu AI connects to the room or the number, drives full multi-turn conversations across personas and edge cases, and scores every run.

Before You Begin

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Before you connect a LiveKit agent, make sure you have:

• A running LiveKit agent, self-hosted or on LiveKit Cloud
• A SIP trunk with a provider such as Twilio or Telnyx, for phone agents only
• TestMu AI workspace with agent-testing permissions
• Source docs for scenario generation: prompt, PRD, or knowledge base

Test a LiveKit Phone Agent

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Phone testing covers inbound and outbound callers arriving over a SIP trunk. Paste the agent's prompt to generate scenarios, then add the number provisioned on the trunk and pick from 100+ voices, background noise conditions, and personas. It covers DTMF entry, SIP REFER transfers, voicemail handling, and carrier latency.

Full setup: Phone agent testing

Test a LiveKit Voice Agent

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Voice testing applies to agents reached by joining a room directly rather than over a phone number. Paste the prompt to generate scenarios, then create the endpoint profile using WebRTC against the room. It isolates the speech pipeline, so STT and TTS accuracy, turn-taking, and interruption handling get tested without telephony noise in the results.

Full setup: Voice agent testing

What You Get With Agent Testing

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Every LiveKit run, on any surface it supports, is scored across:

• 30+ metrics across 8 categories, with configurable thresholds
• Context-aware scenario generation
• Adversarial testing and automated issue detection
• 100+ voices and personas
• Multilingual conversations
• Quality scoring and real-time call monitoring

LiveKit-Specific Considerations

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A few LiveKit behaviours are worth building dedicated scenarios around:

• The unit under test is a room, not an agent ID, so setup differs from every hosted platform
• SIP REFER must be enabled on the trunk for transfers, and Twilio additionally needs PSTN transfer switched on
• Connectors bridge Twilio calls over WebSocket without a SIP trunk, which is a separate connection path
• DTMF, cold transfer, and region pinning are trunk-level settings, so a failure may sit in the trunk rather than the agent
• There is no text channel to test, so the page covers phone and voice only

Troubleshooting

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Common failure modes to watch for:

• Agent not present in the room
• SIP REFER rejected
• DTMF not received
• Codec negotiation failure