Inbound calls
Every number points to a SIP target. We send each inbound call there.
trunkbeam numbers update +493023125417 --sip sip:agent@voice.example
Several targets are tried in order when the first does not answer. The last target may be a regular phone number, for example your office:
trunkbeam numbers update +493023125417 \
--sip sip:agent@voice.example \
--fallback sip:backup@voice.example \
--fallback +4930123456
The SIP INVITE carries the caller’s number in the From header and the dialed number in the To header, so your agent can behave differently per number.
Outbound calls
For outbound calls, create a SIP trunk. You get host, username and password to enter at your voice platform.
trunkbeam trunks create --name vapi-production
Host: sip.trunkbeam.com
Username: trk_4c1e9
Password: (shown once)
Numbers: all numbers in your account
Only numbers from your account are allowed as caller ID. We reject anything else, so nobody can place calls from numbers they do not own.
Setup at your platform
Voice platforms usually call this “Bring your own SIP trunk”, “Custom telephony” or “SIP trunk”. The flow is similar everywhere:
- Create a SIP trunk at the platform with host
sip.trunkbeam.comand the credentials above. - Add your number at the platform and assign it to your agent.
- Set the SIP address the platform gives you for inbound calls as the target with us.
HD voice with G.722
TrunkBeam supports HD voice with the G.722 wideband codec, which carries a wider speech frequency range than traditional narrowband telephony.
HD quality requires your device, voice platform and the entire call path to support wideband audio. The systems on the SIP connection must negotiate a suitable codec. An HD-capable phone alone is not enough if another leg carries only narrowband audio, such as G.711. Converting that audio to G.722 later does not restore the lost frequencies.
This also applies to transfers: HD quality on the new connection depends on the destination device and call path.
Technical background: ITU-T G.722 and SDP codec negotiation in RFC 3264.
Technical details
| Signaling | SIP over TLS (port 5061), UDP and TCP (port 5060) |
| Media | SRTP and RTP |
| Codecs | G.711 A-law (PCMA), G.711 µ-law (PCMU), G.722, Opus |
| DTMF | RFC 2833 (telephone-event) |
| Number format | E.164, for example +493023125417 |
We will publish the IP ranges for firewall rules at launch.