SIP

Run assistant-api with the built-in SIP server for direct SIP connectivity.

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.

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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.

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Firewall Ports

Port	Protocol	Purpose
5090	UDP	SIP signalling
10000–20000	UDP	RTP media

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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)

Example:

sip:123456789:rpd-abc123@203.0.113.10:5090

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Asterisk SIP Trunk (Optional)

To route calls from Asterisk through Rapida over SIP:

pjsip.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}

extensions.conf

; /etc/asterisk/extensions.conf
[rapida-outbound]
exten => _X.,1,NoOp(Route through Rapida SIP)
 same => n,Dial(PJSIP/${EXTEN}@rapida-trunk)
 same => n,Hangup()

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Rapida SaaS SIP Endpoint

For Rapida SaaS, the production SIP server is:

sip:{assistantID}:{apiKey}@sip-01.in.rapida.ai:5060

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Asterisk

Asterisk AudioSocket and WebSocket transports

Configuration

SIP__EXTERNAL_IP and other SIP env vars

Telephony Overview

All providers comparison

Configure Your Own

Add a new telephony provider