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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

PortProtocolPurpose
5090UDPSIP signalling
10000–20000UDPRTP media

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Connecting a SIP Client

Route inbound calls to one of the accepted SIP destination users:
FormatExampleResolution
sip:agent-{assistantID}@{SIP__EXTERNAL_IP}:{SIP__PORT}sip:agent-123456789@203.0.113.10:5090Routes directly by numeric assistant ID
sip:did-{did}@{SIP__EXTERNAL_IP}:{SIP__PORT}sip:did-+15551234567@203.0.113.10:5090Looks up the active SIP phone deployment whose phone value exactly matches {did}
sip:{did}@{SIP__EXTERNAL_IP}:{SIP__PORT}sip:+15551234567@203.0.113.10:5090Same DID lookup, without the did- prefix
The DID value is matched exactly. For example, +15551234567 and 15551234567 are different route values.
Credential-style users such as sip:{assistantID}:{apiKey}@host are not accepted for inbound routing.

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Asterisk SIP Trunk (Optional)

To route calls from Asterisk through Rapida over SIP: 
; /etc/asterisk/pjsip.conf
[rapida-trunk]
type=endpoint
context=rapida-outbound
disallow=all
allow=ulaw
aors=rapida-trunk-aor

[rapida-trunk-aor]
type=aor
contact=sip:{SIP__EXTERNAL_IP}:{SIP__PORT}

[rapida-dialer]
type=identify
endpoint=rapida-trunk
match={SIP__EXTERNAL_IP}

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Rapida SaaS SIP Endpoint

For Rapida SaaS, the production SIP server is: 
sip:agent-{assistantID}@sip-01.in.rapida.ai:5060
You can also route by DID: 
sip:did-{did}@sip-01.in.rapida.ai:5060
sip:{did}@sip-01.in.rapida.ai:5060

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