SIP Trunking & Numbers
SIP trunking connects a phone system or application to the public telephone network over IP, replacing physical lines with channels of concurrent calls. SIP.IO provides first-class SIP trunks for bring-your-own-carrier (BYOC) and connecting a customer PBX, plus a global number inventory, outbound routing, and number transforms, secured with TLS signaling and SRTP media.
What you can do
- Bring your own carrier (BYOC): connect an existing carrier or wholesale trunk and keep your rates.
- Connect a PBX. Send calls from a customer PBX to the platform, and deliver inbound DIDs back to the PBX (receive-on-PBX).
- Provision numbers: local DIDs and global termination with per-country caller-ID.
- Route outbound: match dialed numbers to trunks or the default carrier by prefix/country and priority.
- Transform numbers. Regex rewrites for dialed number and caller-ID at account, trunk, DID, or route scope.
Trunk model
A single first-class trunk model spans carrier, BYOC, and customer-PBX peers, in either direction, with IP, registration, or digest authentication and registration failover across nodes. See Trunks, Outbound & PSTN for the full reference.
| Capability | Docs |
|---|---|
| SIP trunks (carrier / BYOC / PBX) | Trunks, Outbound & PSTN |
| Numbers & extensions, routing | Numbers & Extensions |
| Outbound routes & number transforms | Outbound |
| Concurrency / channel limits | Concurrency Control |
| Caller-ID presentation | Caller-ID |
Secure by default
Signaling rides TLS and audio rides SRTP, with per-tenant isolation; encryption isn’t an add-on.
FAQ
What is SIP trunking? Connecting a phone system/app to the phone network over IP via SIP, using channels instead of physical lines.
BYOC supported? Yes: first-class trunks for your own carrier or a customer PBX, with IP/registration/digest auth, secured by TLS/SRTP.
Bring my own numbers? Provision DIDs on SIP.IO or connect carrier numbers via a BYOC trunk.