SIP Trunk

SIP (Session Initiation Protocol) Trunking provides a direct connection between your telephony infrastructure and Rapida. This is ideal for organizations that already use a SIP-compatible PBX (such as FreeSWITCH, Kamailio, or hosted SIP providers) and want to connect their existing phone lines to an AI voice assistant without relying on a specific cloud telephony vendor.

SIP Trunk is a bring-your-own-carrier option. Rapida runs a native SIP server that accepts inbound SIP INVITE requests and establishes RTP media sessions directly. This provides the lowest latency path for voice AI interactions.

How It Works

Prerequisites

SIP Infrastructure

A SIP-compatible PBX, carrier, or VoIP provider that can send calls to an external SIP URI

Rapida Account

An active Rapida account with a configured voice assistant

You will need:

Your SIP trunk provider’s SIP URI (e.g., sip:trunk.provider.com:5060)
Optional SIP Username and Password if your trunk requires authentication

Step 1: Set Up Provider Credentials

Store your SIP trunk credentials in Rapida so the platform can authenticate with your SIP infrastructure.

Navigate to External Integrations

Go to Integration > Tools in the Rapida dashboard. You will see a grid of available external integrations.

Select SIP Trunk

Find the SIP Trunk card and click “Setup Credential”.

Enter Your SIP Credentials

A modal will appear. Fill in the following fields:

Field	Description	Required
Key Name	A friendly name for this credential (e.g., “Office PBX”)	Yes
SIP URI	The SIP trunk endpoint (e.g., `sip:trunk.provider.com:5060`)	Yes
SIP Username	Authentication username for the SIP trunk	No
SIP Password	Authentication password for the SIP trunk	No

Click “Configure” to save.

If your SIP provider uses IP-based authentication (allow-listing Rapida’s IP), you can leave the username and password fields empty.

Verify Connection

After saving, the SIP Trunk card should display “Connected”. Click on it to see credential details, last updated time, and management options.

Step 2: Configure Phone Deployment

With credentials saved, configure your assistant’s phone deployment to use SIP.

Open Your Assistant

Navigate to Assistants and select the assistant you want to deploy via phone.

Go to Phone Deployment

Click “Deploy” → “Phone” to open the phone deployment configuration page.

Select SIP Trunk as Telephony Provider

In the Telephony section:

Select SIP Trunk from the telephony provider dropdown
Choose the SIP credential you created in Step 1 from the Credential dropdown
Enter the Caller ID — the phone number to display for outbound calls (e.g., +15551234567)

Configure Experience Settings

Set up the conversation experience:

Setting	Description	Default
Greeting	The first message the AI speaks when answering	(optional)
Error Message	Message spoken when an error occurs	(optional)
Idle Timeout	Seconds before prompting an idle caller	`30`
Idle Message	Message spoken when caller is idle	`"Are you there?"`
Idle Timeout Retries	How many times to retry before ending call	`2`
Max Call Duration	Maximum call length in seconds	`300`

Configure Audio Providers

Select your Speech-to-Text (STT) and Text-to-Speech (TTS) providers. These determine how audio is transcribed and synthesized during the call.

Save Deployment

Click “Deploy” to save. Your assistant is now ready to handle phone calls via SIP.

Step 3: Configure Your PBX / SIP Provider

Route calls from your PBX or SIP carrier to Rapida’s SIP server.

Create a SIP Trunk in Your PBX

In your PBX admin panel (FreeSWITCH, Asterisk, Kamailio, or hosted provider), create a new SIP trunk pointing to Rapida’s SIP endpoint.Rapida SIP Server:

sip:sip-01.in.rapida.ai:5060

Configure your trunk with:

Outbound Proxy: sip-01.in.rapida.ai:5060
Transport: UDP or TCP
Codec: G.711 μ-law (PCMU) preferred, G.711 A-law (PCMA) supported

Create a Dial Plan Rule

Add a dial plan entry that routes specific numbers or patterns to the Rapida SIP trunk.FreeSWITCH example:

<extension name="rapida-ai">
  <condition field="destination_number" expression="^(5551234567)$">
    <action application="bridge" data="sofia/external/sip:did-$1@sip-01.in.rapida.ai:5060"/>
  </condition>
</extension>

Asterisk example (extensions.conf):

[rapida-trunk]
exten => _X.,1,Dial(SIP/did-${EXTEN}@rapida-trunk)

Verify Routing

Place a test call to confirm the call reaches Rapida and the AI assistant answers.

Inbound SIP URI Routing

Rapida routes inbound SIP calls from the user portion of the request URI. The To URI is used first; if it has no user, Rapida falls back to the From URI. Accepted inbound destination formats:

Format	Example	Resolution
`sip:agent-{assistantID}@{host}:{port}`	`sip:agent-123456789@sip-01.in.rapida.ai:5060`	Routes directly by numeric assistant ID
`sip:did-{did}@{host}:{port}`	`sip:did-+15551234567@sip-01.in.rapida.ai:5060`	Looks up the active SIP phone deployment whose phone value exactly matches `{did}`
`sip:{did}@{host}:{port}`	`sip:+15551234567@sip-01.in.rapida.ai:5060`	Same DID lookup, without the `did-` prefix

The same route user forms are accepted with sips: when the SIP listener is configured for TLS.

The DID value is matched exactly against the phone number stored on the assistant’s active SIP phone deployment. +15551234567 and 15551234567 are different route values.

Credential-style users such as sip:{assistantID}:{apiKey}@host are not accepted for inbound routing. Use agent-{assistantID}, did-{did}, or the raw DID form.

Making Outbound Calls

Once your phone deployment is configured, you can initiate outbound calls through your SIP trunk using the Rapida API or SDKs.

Python
Node.js
cURL

from rapida import Rapida

client = Rapida(api_key="rpd-xxx-your-key")

call = client.calls.create(
    assistant_id=123456789,
    to_number="+15551234567",
    metadata={"campaign": "follow-up"},
)

print(f"Call queued: conversation_id={call.conversation.id}")

import { Rapida } from 'rapida';

const client = new Rapida({ apiKey: 'rpd-xxx-your-key' });

const call = await client.calls.create({
  assistantId: 123456789,
  toNumber: '+15551234567',
  metadata: { campaign: 'follow-up' },
});

console.log(`Call queued: ${call.conversation.id}`);

curl -X POST https://api.rapida.ai/v1/talk/call \
  -H "Authorization: Bearer rpd-xxx-your-key" \
  -H "Content-Type: application/json" \
  -d '{
    "assistant": { "assistant_id": 123456789 },
    "to_number": "+15551234567",
    "metadata": { "campaign": "follow-up" }
  }'

Features

Feature	Description
Bring Your Own Carrier	Use any SIP-compatible provider or on-premise PBX
Direct SIP Connection	Native SIP INVITE/BYE handling with RTP media — no intermediary cloud provider
Low Latency	Direct RTP audio path minimizes round-trip delay
Inbound Calls	Accept calls routed from your PBX to the AI assistant
Outbound Calls	Initiate calls programmatically via SDK or API through your SIP trunk
Call Recording	Automatic conversation capture for review and compliance
Codec Support	G.711 μ-law (PCMU), G.711 A-law (PCMA)
Flexible Auth	Digest authentication or IP-based allow-listing

Supported PBX / Providers

SIP Trunk works with any standard SIP-compliant system, including:

System	Type
FreeSWITCH	Open-source softswitch
Asterisk	Open-source PBX (also has dedicated integration)
Kamailio	SIP proxy/router
Twilio Elastic SIP	Cloud SIP trunking
Vonage SIP Connect	Cloud SIP trunking
Bandwidth	Cloud SIP trunking
Telnyx	Cloud SIP trunking
SignalWire	Cloud SIP trunking

If you’re using Twilio, Vonage, Exotel, or Asterisk, consider using the dedicated integrations for those providers — they offer additional features like webhook-based call control and provider-specific optimizations.

Troubleshooting

SIP INVITE rejected or no response

Verify the SIP URI in your credential is correct and reachable
Check that your PBX can route to sip-01.in.rapida.ai:5060 (firewall/NAT rules)
If using authentication, ensure username and password are correct
Try both UDP and TCP transport if one doesn’t work

One-way audio or no audio

Check RTP port ranges — ensure your firewall allows UDP traffic for RTP media
Verify codec negotiation — Rapida supports G.711 μ-law (PCMU) and G.711 A-law (PCMA)
If behind NAT, ensure your PBX has proper STUN/TURN or external media IP configuration
Verify STT and TTS providers are configured in the phone deployment

Outbound calls fail

Verify the SIP credential is connected in Integration > Tools
Ensure the phone deployment is configured and saved
Check that your SIP trunk allows outbound calls from Rapida’s IP

Audio quality issues

Use a wired network connection for your PBX when possible
Prefer G.711 μ-law (PCMU) codec for compatibility
Minimize network hops between your PBX and Rapida
Check for packet loss or jitter on the RTP path using network monitoring tools

Create an Assistant

Build your voice AI assistant

Phone Deployment

Overview of phone deployment options

Asterisk Integration

Dedicated Asterisk PBX integration with AudioSocket

Conversation Logs

View call history and transcripts

Assistants

Knowledge

LLM Endpoint

Activity & Logs

External Integrations

Credentials

Workspace

Governance

Deployment Options

How It Works

Prerequisites

SIP Infrastructure

Rapida Account

Step 1: Set Up Provider Credentials

Step 2: Configure Phone Deployment

Step 3: Configure Your PBX / SIP Provider

Inbound SIP URI Routing

Making Outbound Calls

Features

Supported PBX / Providers

Troubleshooting

Create an Assistant

Phone Deployment

Asterisk Integration

Conversation Logs

​How It Works

​Prerequisites

SIP Infrastructure

Rapida Account

​Step 1: Set Up Provider Credentials

​Step 2: Configure Phone Deployment

​Step 3: Configure Your PBX / SIP Provider

​Inbound SIP URI Routing

​Making Outbound Calls

​Features

​Supported PBX / Providers

​Troubleshooting

​Related Resources

Create an Assistant

Phone Deployment

Asterisk Integration

Conversation Logs

How It Works

Prerequisites

Step 1: Set Up Provider Credentials

Step 2: Configure Phone Deployment

Step 3: Configure Your PBX / SIP Provider

Inbound SIP URI Routing

Making Outbound Calls

Features

Supported PBX / Providers

Troubleshooting

Related Resources