SIP for Voice Connections
Call your voice agent's SIP address with digest authentication, custom headers and SRTP media encryption.
Overview
Use the SIP transport when your voice agent answers a SIP address, for example on your own PBX, a SIP-enabled contact center, or a voice platform that gives you a SIP URI. Confident AI sends an INVITE straight to that address, skipping the phone network.
SIP Address
Enter the address your agent answers on, in the form sip:user@host, for example sip:agent@sip.example.com. Include a port if your server doesn't listen on 5060 (for example sip:agent@sip.example.com:5070).
Signaling Transport
Pick how the SIP signaling reaches your server: UDP (default), TCP, or TLS.
A sips: address always connects over TLS, and a ;transport= parameter on the address (udp, tcp, or tls) takes precedence over the selected tab.
Media Encryption
Media encryption controls whether the call audio is encrypted with SRTP:
| Option | Description |
|---|---|
| Disabled (default) | Audio is sent as plain RTP. |
| Allowed | Confident AI offers SRTP and falls back to plain audio when your side can't encrypt. |
| Required | Confident AI only accepts SRTP. The call fails instead of falling back when your side can't encrypt. |
Choose Required when your server rejects unencrypted media, and Allowed when you'd like encryption but some of your endpoints don't support it yet.
Digest Authentication
Turn on Digest authentication if your server challenges the INVITE, and enter the Username and Password it expects. Leave it off when your server accepts calls without credentials or allowlists by IP.
Caller ID
Caller ID (optional) is sent as the From header of the INVITE, as a phone number or a SIP URI. Leave it blank to use Confident AI's default.
Custom Headers
Custom headers (optional) are added to the INVITE. Use them to route the call to a specific agent or tenant, for example X-Agent-Id: support-en.
Hanging Up
Confident AI hangs up the call with a SIP BYE as soon as the conversation is over, or when the call reaches the connection's Max call length or Silence timeout. Your agent can also hang up first by sending its own BYE, which ends the conversation at that turn.
Make sure your agent releases the call when it receives the BYE, so each simulated conversation frees its line before the next one starts.
Testing Your Connection
Click Dial in the Test call console. The console shows Dialing… and then Ringing… while it waits for your agent to answer, for up to the connection's Ring timeout. See Testing Your Connection for what each result means.
Next Steps
WebRTC
Join your agent through LiveKit, Pipecat, or your own signaling URL.
Multi-Turn Evals
Run multi-turn evaluations against your voice connection.
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