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Documentation Index

Fetch the complete documentation index at: https://docs.usetuner.ai/llms.txt

Use this file to discover all available pages before exploring further.

Prerequisite: your agent must already be integrated with Tuner and successfully sending real calls before you configure simulation. Simulation reuses the same Evals, Intents, and Call Outcomes your live agent is wired to, so that integration has to be in place first. Set it up here:
To run simulations, Tuner places SIP calls to your agent. Agent Settings → SIP Settings needs a SIP URI (for example sip:agent@sip.vapi.ai) and, when your SIP edge uses digest authentication, a username and password. Tuner does not care which vendor supplies those values; they must match whatever your SIP ingress expects. Some stacks only expose a URI (leave username and password empty). Tuner Agent Settings — SIP Settings (SIP URI, username, password)

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What is SIP, and why does Tuner need it?

SIP (Session Initiation Protocol, RFC 3261) is the signaling protocol voice platforms use to start, manage, and end a call, every voice agent stack speaks SIP at its ingress. Tuner uses SIP so simulations enter your agent through the same path a real phone call would, same audio pipeline, same transcription, same prompt, same tools. That’s why we ask for a SIP URI (where to dial) and, if your edge uses digest auth, a username and password.

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How a simulation call actually flows

When you click Run Simulation, this happens under the hood:
  1. Tuner builds the scenario, picks a caller persona, an intent, and what to stress-test.
  2. Tuner sends a SIP INVITE to the SIP URI you pasted. This is the “ring”, a small text message over the internet that says “I want to start a call with you.” That INVITE carries a SIP Call-ID Tuner generated for this simulation — it’s the key that links what Tuner dialled out with what your agent syncs back later.
  3. Your SIP edge responds. If digest auth is on, it replies with a 407 Proxy Authentication Required challenge. Tuner re-sends the INVITE with the username/password you saved. If credentials match, the edge accepts and forwards the call to your agent.
  4. Audio flows. Once the call is up, Tuner streams the simulated caller’s voice (RTP) to your agent, and your agent’s responses stream back. From your agent’s perspective this is a normal inbound phone call.
  5. Your agent syncs the call back to Tuner with the SIP Call-ID. When the call ends, your integration posts the transcript and metadata back to Tuner, with the SIP Call-ID included. If you’re on Vapi or Retell, we handle this for you out of the box, no work on your side. LiveKit and Pipecat integrations read the SIP Call-ID from the inbound call and pass it through the SDK’s plugin/observer. Custom integrations include sip_call_id in their Create Call API payload directly. That value is how Tuner knows this synced call is the one it just simulated, rather than ordinary production traffic.
  6. Evaluation runs. Once the synced call is matched to the simulation, Tuner records the transcript, latency, tool traces, and runs your production Evals against it.
You never see steps 2–3, they happen in milliseconds. But this is why the URI and credentials have to be exactly right: a typo means the INVITE lands at the wrong host, or the 407 challenge fails, and the call never connects. And without sip_call_id coming back in step 5, the call still happens, it just shows up as a normal production call instead of a simulation result.

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Anatomy of a SIP URI

sip:user-part@host
  • sip:, the scheme (like https:// for web URLs). You may rarely see sips: for SIP over TLS; paste it verbatim.
  • user-part, an identifier your provider assigns (Vapi’s SIP Identifier, Twilio/Telnyx credential username). Sometimes absent, the host alone is enough.
  • host, the public address of your SIP edge (e.g. sip.vapi.ai, sip.telnyx.com, your-project.sip.livekit.cloud).
A URI may also carry parameters after the host (e.g. ;transport=tcp). Copy whatever your provider’s console shows, verbatim.

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Choose your platform

Pick where your SIP URI and credentials come from:

SIP for Vapi

Free Vapi SIP trunk: URI + digest credentials.

SIP for Retell

SIP URI from Retell for your agent (often URI only).

SIP for other platforms

Twilio or Telnyx trunks, LiveKit or Pipecat agents, or any SIP URI you already have.
Paste your SIP URI and credentials into Agent Settings → SIP Settings, click Verify & Save, and you’re ready to run simulations.
Next: Run your first simulation.