Connect your telephony (SIP).
gptagent runs on the phone line you already have. Point your SIP trunk at us — or let us register to your carrier — and your AI agents answer inbound calls straight away. No new phone stack, no porting your numbers.
Pick the model that matches your carrier.
Both work the same for callers. Choose whichever your telephony provider supports — we'll set our side up to match.
We register to your carrier
We sign in to your SIP provider with the credentials you issue us — the same way a desk phone or softphone would. Calls to your number then flow to your gptagent agents. This is the quickest option if your provider gives out SIP logins (many hosted providers do).
- A SIP username and password for a trunk or line
- Your registration server or domain (for example, the host your softphones connect to)
You send calls to us
Your telephony system sends calls to us from a fixed IP address, and we recognise your traffic by that address — no login required. This suits carriers and on-premise PBXs that connect trunks by IP rather than by registration.
- The IP address(es) your signalling comes from
- The IP address(es) or ranges your voice (media) comes from — these are often different, so please include both
Encrypted, and locked to your addresses.
Encryption end to end of the connection
We support TLS for the call signalling and SRTP for the voice itself, so both the setup of the call and the audio are encrypted in transit. Plain (unencrypted) UDP and TCP are also supported if your carrier needs them — but we recommend TLS with SRTP.
No certificate to install
You don't need to obtain or install any certificate on your side to use the encrypted connection. We handle the certificate on ours.
Only your traffic gets in
The connection accepts calls only from the addresses you tell us about. Everything else is blocked at the edge — so random internet scanners can't reach your line.
The settings sheet for your telephony admin.
| Signalling host | Your gptagent connection address — we confirm it for your account |
|---|---|
| Ports | 5060 (UDP or TCP) · 5061 (TLS / encrypted) |
| Transport | UDP, TCP, or TLS — TLS recommended |
| Voice encryption | SRTP over TLS (recommended); plain RTP over UDP/TCP also supported |
| Codecs | G.711 (µ-law / A-law) |
| Voice (RTP) ports | 10000–10100 (UDP) |
| Connection model | Outbound registration (we register to you) or inbound IP trunk (you send from a fixed IP) |
Give this to whoever manages your SIP trunk. We'll confirm the exact host address for your account when you connect.
A short checklist to get started.
If we register to your carrier
- SIP username and password
- Registration server or domain
- Preferred transport (UDP, TCP or TLS)
If you send calls to us (IP trunk)
- Your signalling source IP address(es)
- Your media source IP address(es) or ranges
- Preferred transport (UDP, TCP or TLS)
Not sure which fits your setup?
Tell us who your telephony provider is and we'll tell you exactly how to connect — and set it up with you on a short call.
Talk to us