The assistant-api includes a built-in SIP server. Any SIP client, softphone, or PBX can call it directly — no external telephony account is required.
Required Configuration
# docker/assistant-api/.assistant.env
SIP__SERVER=0.0.0.0 # Bind address (listen on all interfaces)
SIP__EXTERNAL_IP=203.0.113.10 # Your server's PUBLIC IP — advertised in SDP
SIP__PORT=5090 # SIP signalling port (UDP)
SIP__RTP_PORT_RANGE_START=10000
SIP__RTP_PORT_RANGE_END=20000
SIP__EXTERNAL_IP must be your server’s real public IP. If left as 0.0.0.0, SDP will advertise a non-routable address, breaking the RTP media path — callers will hear silence.
Firewall Ports
| Port | Protocol | Purpose |
|---|
5090 | UDP | SIP signalling |
10000–20000 | UDP | RTP media |
Connecting a SIP Client
Use this connection string from any SIP phone, softphone (e.g. Zoiper, Linphone), or PBX:
sip:{assistantID}:{apiKey}@{SIP__EXTERNAL_IP}:{SIP__PORT}
| Field | Value |
|---|
assistantID | Numeric assistant ID from the Rapida dashboard |
apiKey | Rapida API key for the project (rpd-xxx) |
SIP__EXTERNAL_IP | Public IP set in env var |
SIP__PORT | 5090 (default) |
sip:123456789:rpd-abc123@203.0.113.10:5090
Asterisk SIP Trunk (Optional)
To route calls from Asterisk through Rapida over SIP:
pjsip.conf
extensions.conf
; /etc/asterisk/pjsip.conf
[rapida-trunk]
type=endpoint
context=rapida-outbound
disallow=all
allow=ulaw
aors=rapida-trunk-aor
outbound_auth=rapida-trunk-auth
[rapida-trunk-auth]
type=auth
auth_type=userpass
username={assistantID}
password={apiKey}
[rapida-trunk-aor]
type=aor
contact=sip:{SIP__EXTERNAL_IP}:{SIP__PORT}
[rapida-dialer]
type=identify
endpoint=rapida-trunk
match={SIP__EXTERNAL_IP}
; /etc/asterisk/extensions.conf
[rapida-outbound]
exten => _X.,1,NoOp(Route through Rapida SIP)
same => n,Dial(PJSIP/${EXTEN}@rapida-trunk)
same => n,Hangup()
Rapida SaaS SIP Endpoint
For Rapida SaaS, the production SIP server is:
sip:{assistantID}:{apiKey}@sip-01.in.rapida.ai:5060
