SIP

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

Route inbound calls to one of the accepted SIP destination users:

Format	Example	Resolution
`sip:agent-{assistantID}@{SIP__EXTERNAL_IP}:{SIP__PORT}`	`sip:agent-123456789@203.0.113.10:5090`	Routes directly by numeric assistant ID
`sip:did-{did}@{SIP__EXTERNAL_IP}:{SIP__PORT}`	`sip:did-+15551234567@203.0.113.10:5090`	Looks up the active SIP phone deployment whose phone value exactly matches `{did}`
`sip:{did}@{SIP__EXTERNAL_IP}:{SIP__PORT}`	`sip:+15551234567@203.0.113.10:5090`	Same 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.

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

[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/did-${EXTEN}@rapida-trunk)
 same => n,Hangup()

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

Getting Started

Dashboard (UI)

Web API

Assistant API

Integration API

Endpoint API

Document API

Required Configuration

Firewall Ports

Connecting a SIP Client

Asterisk SIP Trunk (Optional)

Rapida SaaS SIP Endpoint

Asterisk

Configuration

Telephony Overview

Configure Your Own

​Required Configuration

​Firewall Ports

​Connecting a SIP Client

​Asterisk SIP Trunk (Optional)

​Rapida SaaS SIP Endpoint

Asterisk

Configuration

Telephony Overview

Configure Your Own

Required Configuration

Firewall Ports

Connecting a SIP Client

Asterisk SIP Trunk (Optional)

Rapida SaaS SIP Endpoint