Voice AI Integrations

LiveKit–IraVoice Integration

IraVoice enables LiveKit integration by proxying inbound and outbound PSTN calls between LiveKit rooms and telecom networks.

This guide explains how to enable inbound and outbound PSTN calling between LiveKit rooms and the public telephone network using the IraVoice platform and Epicode SIP connectivity.

Architecture Overview

Outbound Call Flow

Inbound Call Flow

SIP Configuration

The following SIP credentials will be provided by the Epicode team and must be used while creating SIP trunks in LiveKit.

Parameter	Description
SIP Server IP	Provided by Epicode
SIP Port	Provided by Epicode
SIP Username	Provided by Epicode
SIP Password	Provided by Epicode
Allowed Numbers (DIDs / Caller IDs)	Provided by Epicode

Outbound Calling Setup (LiveKit → PSTN)

Step 1: Create an Outbound SIP Trunk

Create an outbound SIP trunk in LiveKit using the SIP credentials provided by Epicode.

Reference: https://docs.livekit.io/telephony/making-calls/outbound-trunk/

Example


Code
{
  "trunk": {
    "name": "SL outbound trunk",
    "address": "111.12.128.69:5090",
    "numbers": ["+918044781234"],
    "auth_username": "1019",
    "auth_password": "pass@123"
  }
}

Step 2: Create a SIP Participant

From your application, create a SIP participant in a LiveKit room with:

Destination phone number
Room name
SIP trunk ID
Custom SIP X-* headers (optional)

Reference: https://docs.livekit.io/telephony/making-calls/outbound-calls/

Important Notes

Prefix the destination number with 789 (or another mutually agreed prefix) before placing the call.
This prefix is used by the Epicode dialplan to identify and correctly route calls originating from your LiveKit deployment.
Custom SIP X-* headers can be passed and will be forwarded to IraVoice.

Example


Code
{
  "sip_number": "+918044781234",
  "sip_trunk_id": "ST_ebQsvbf9TgC2",
  "sip_call_to": "7896366430222",
  "room_name": "open-room",
  "participant_identity": "sip-test",
  "participant_name": "Test call participant",
  "wait_until_answered": true,
  "headers": {
    "X-Livekit-ID": "livekit-id"
  }
}

Result

The destination phone user joins the LiveKit room as a participant.

Inbound Calling Setup (PSTN → LiveKit)

LiveKit SIP Endpoint

To enable inbound calling, share your LiveKit SIP endpoint with the Epicode team.

Endpoint Format

Code
{destination_number}@abcdefghijk.sip.livekit.cloud;transport=tcp

Example

Code
+919876543210@kejbcnkesbvb.sip.livekit.cloud;transport=tcp

This endpoint will be configured by Epicode as the SIP destination for inbound calls.

Step 1: Configure an Inbound SIP Trunk

Create an inbound SIP trunk in LiveKit and configure the DID numbers that will receive calls.

Reference: https://docs.livekit.io/telephony/accepting-calls/inbound-trunk/

Example


Code
{
  "trunk": {
    "name": "My trunk",
    "numbers": ["+919876595372"],
    "krisp_enabled": false
  }
}

Step 2: Create a Dispatch Rule

Create a dispatch rule that routes incoming calls to a specific LiveKit room.

Reference: https://docs.livekit.io/telephony/accepting-calls/inbound-trunk/

Example


Code
{
  "dispatch_rule": {
    "rule": {
      "dispatchRuleDirect": {
        "roomName": "open-room"
      }
    },
    "name": "My dispatch rule"
  }
}

Result

When a user calls the configured DID, the call is automatically connected to the mapped LiveKit room.

Call Recording

Epicode supports multiple recording delivery options through the IraVoice platform:

Push recordings to a customer-provided HTTP endpoint (webhook).
Upload recordings directly to customer-managed cloud storage.

Supported storage platforms include:

Amazon S3
Google Cloud Storage (GCS)
Microsoft Azure Blob Storage

Recording delivery can be configured independently for test and production environments.

Real-Time Call Events

The IraVoice platform can send real-time call lifecycle events to a customer-provided HTTP endpoint (webhook).

Supported events include:

Event
Ringing
Answered
Bridged
Unbridged
Hangup
Call Quality Metrics

Sample Ringing Event


Code
{
  "event_name": "iravoice::ringing",
  "event_data": {
    "call_uuid": "cd94574d-e51a-4689-80ad-daf753453ce9",
    "other_leg_uuid": "799c937b-b34d-47fc-8238-1088439a1ad0",
    "network_ip": "10.5.110.117",
    "tenant_id": "xyz",
    "call_type": "proxy",
    "to_number": "+911234567899",
    "from_number": "+911234567890",
    "call_params": {
      "call_direction": "outbound",
      "event_url": "http://192.168.1.5:8003/epicode-logs",
      "recording_format": "wav"
    },
    "sip_fields": {
      "livekit_id": "cf6d8581-7128-4dd7-9ce5-8f6871057626"
    },
    "direction": "outbound",
    "timestamp": "2026-02-18T11:59:43.149058Z"
  }
}

Sample Answered Event


Code
{
  "event_name": "iravoice::answered",
  "event_data": {
    "tenant_id": "xyz",
    "call_uuid": "799c937b-b34d-47fc-8238-1088439a1ad0",
    "call_params": {
      "call_direction": "outbound",
      "event_url": "http://192.168.1.5:8003/epicode-logs",
      "recording_format": "wav"
    },
    "dialer_ip": "192.168.11.13",
    "gateway": "",
    "call_type": "proxy",
    "direction": "inbound",
    "network_ip": "127.0.0.1",
    "timestamp": "2026-02-18T11:59:43.160396Z",
    "sip_fields": {
      "livekit_id": "cf6d8581-7128-4dd7-9ce5-8f6871057626"
    }
  }
}

Sample Hangup Event


Code
{
  "event_name": "iravoice::hangup",
  "event_data": {
    "tenant_id": "xyz",
    "call_uuid": "62b0ab5e-483d-4e61-8143-ecde97e36d23",
    "call_params": {
      "call_direction": "outbound",
      "event_url": "http://192.168.1.5:8003/epicode-logs",
      "recording_format": "wav"
    },
    "dialer_ip": "192.168.11.13",
    "gateway": "",
    "call_type": "proxy",
    "direction": "inbound",
    "network_ip": "127.0.0.1",
    "failure_status": "480",
    "failure_phrase": "Temporarily not available",
    "from_number": "+917123451234",
    "to_number": "1234567890",
    "hangup_cause": "NORMAL_CLEARING",
    "sip_fields": {
      "livekit_id": "c12ac441-399c-45d8-bc99-ae187cc3bfcc"
    },
    "timestamp": "2026-02-18T11:59:42.466362Z"
  }
}

Sample Call Quality Event


Code
{
    "event_name": "iravoice::call_quality",
    "event_data": {
        "tenant_id": "xyz",
        "call_uuid": "62b0ab5e-483d-4e61-8143-ecde97e36d23",
        "call_params": {
            "call_direction": "outbound"
        },
        "dialer_ip": "192.168.11.13",
        "gateway": "",
        "call_type": "proxy",
        "direction": "inbound",
        "network_ip": "127.0.0.1",
        "mos": "4.50",
        "quality": "100.00",
        "sip_fields": {
          "livekit_id": "c12ac441-399c-45d8-bc99-ae187cc3bfcc"
        },
        "timestamp": "2026-02-18T11:59:42.765432Z"
    }
}

Support

For SIP credentials, routing configuration, recording setup, webhook integration, or troubleshooting assistance, please contact the Epicode Support Team.

Last modified on June 30, 2026

Server Monitor Vapi

Voice AI Integrations

LiveKit–IraVoice Integration

IraVoice enables LiveKit integration by proxying inbound and outbound PSTN calls between LiveKit rooms and telecom networks.

This guide explains how to enable inbound and outbound PSTN calling between LiveKit rooms and the public telephone network using the IraVoice platform and Epicode SIP connectivity.

Architecture Overview

Outbound Call Flow

Inbound Call Flow

SIP Configuration

The following SIP credentials will be provided by the Epicode team and must be used while creating SIP trunks in LiveKit.

Parameter	Description
SIP Server IP	Provided by Epicode
SIP Port	Provided by Epicode
SIP Username	Provided by Epicode
SIP Password	Provided by Epicode
Allowed Numbers (DIDs / Caller IDs)	Provided by Epicode

Outbound Calling Setup (LiveKit → PSTN)

Step 1: Create an Outbound SIP Trunk

Create an outbound SIP trunk in LiveKit using the SIP credentials provided by Epicode.

Reference: https://docs.livekit.io/telephony/making-calls/outbound-trunk/

Example


Code
{
  "trunk": {
    "name": "SL outbound trunk",
    "address": "111.12.128.69:5090",
    "numbers": ["+918044781234"],
    "auth_username": "1019",
    "auth_password": "pass@123"
  }
}

Step 2: Create a SIP Participant

From your application, create a SIP participant in a LiveKit room with:

Destination phone number
Room name
SIP trunk ID
Custom SIP X-* headers (optional)

Reference: https://docs.livekit.io/telephony/making-calls/outbound-calls/

Important Notes

Prefix the destination number with 789 (or another mutually agreed prefix) before placing the call.
This prefix is used by the Epicode dialplan to identify and correctly route calls originating from your LiveKit deployment.
Custom SIP X-* headers can be passed and will be forwarded to IraVoice.

Example


Code
{
  "sip_number": "+918044781234",
  "sip_trunk_id": "ST_ebQsvbf9TgC2",
  "sip_call_to": "7896366430222",
  "room_name": "open-room",
  "participant_identity": "sip-test",
  "participant_name": "Test call participant",
  "wait_until_answered": true,
  "headers": {
    "X-Livekit-ID": "livekit-id"
  }
}

Result

The destination phone user joins the LiveKit room as a participant.

Inbound Calling Setup (PSTN → LiveKit)

LiveKit SIP Endpoint

To enable inbound calling, share your LiveKit SIP endpoint with the Epicode team.

Endpoint Format

Code
{destination_number}@abcdefghijk.sip.livekit.cloud;transport=tcp

Example

Code
+919876543210@kejbcnkesbvb.sip.livekit.cloud;transport=tcp

This endpoint will be configured by Epicode as the SIP destination for inbound calls.

Step 1: Configure an Inbound SIP Trunk

Create an inbound SIP trunk in LiveKit and configure the DID numbers that will receive calls.

Reference: https://docs.livekit.io/telephony/accepting-calls/inbound-trunk/

Example


Code
{
  "trunk": {
    "name": "My trunk",
    "numbers": ["+919876595372"],
    "krisp_enabled": false
  }
}

Step 2: Create a Dispatch Rule

Create a dispatch rule that routes incoming calls to a specific LiveKit room.

Reference: https://docs.livekit.io/telephony/accepting-calls/inbound-trunk/

Example


Code
{
  "dispatch_rule": {
    "rule": {
      "dispatchRuleDirect": {
        "roomName": "open-room"
      }
    },
    "name": "My dispatch rule"
  }
}

Result

When a user calls the configured DID, the call is automatically connected to the mapped LiveKit room.

Call Recording

Epicode supports multiple recording delivery options through the IraVoice platform:

Push recordings to a customer-provided HTTP endpoint (webhook).
Upload recordings directly to customer-managed cloud storage.

Supported storage platforms include:

Amazon S3
Google Cloud Storage (GCS)
Microsoft Azure Blob Storage

Recording delivery can be configured independently for test and production environments.

Real-Time Call Events

The IraVoice platform can send real-time call lifecycle events to a customer-provided HTTP endpoint (webhook).

Supported events include:

Event
Ringing
Answered
Bridged
Unbridged
Hangup
Call Quality Metrics

Sample Ringing Event


Code
{
  "event_name": "iravoice::ringing",
  "event_data": {
    "call_uuid": "cd94574d-e51a-4689-80ad-daf753453ce9",
    "other_leg_uuid": "799c937b-b34d-47fc-8238-1088439a1ad0",
    "network_ip": "10.5.110.117",
    "tenant_id": "xyz",
    "call_type": "proxy",
    "to_number": "+911234567899",
    "from_number": "+911234567890",
    "call_params": {
      "call_direction": "outbound",
      "event_url": "http://192.168.1.5:8003/epicode-logs",
      "recording_format": "wav"
    },
    "sip_fields": {
      "livekit_id": "cf6d8581-7128-4dd7-9ce5-8f6871057626"
    },
    "direction": "outbound",
    "timestamp": "2026-02-18T11:59:43.149058Z"
  }
}

Sample Answered Event


Code
{
  "event_name": "iravoice::answered",
  "event_data": {
    "tenant_id": "xyz",
    "call_uuid": "799c937b-b34d-47fc-8238-1088439a1ad0",
    "call_params": {
      "call_direction": "outbound",
      "event_url": "http://192.168.1.5:8003/epicode-logs",
      "recording_format": "wav"
    },
    "dialer_ip": "192.168.11.13",
    "gateway": "",
    "call_type": "proxy",
    "direction": "inbound",
    "network_ip": "127.0.0.1",
    "timestamp": "2026-02-18T11:59:43.160396Z",
    "sip_fields": {
      "livekit_id": "cf6d8581-7128-4dd7-9ce5-8f6871057626"
    }
  }
}

Sample Hangup Event


Code
{
  "event_name": "iravoice::hangup",
  "event_data": {
    "tenant_id": "xyz",
    "call_uuid": "62b0ab5e-483d-4e61-8143-ecde97e36d23",
    "call_params": {
      "call_direction": "outbound",
      "event_url": "http://192.168.1.5:8003/epicode-logs",
      "recording_format": "wav"
    },
    "dialer_ip": "192.168.11.13",
    "gateway": "",
    "call_type": "proxy",
    "direction": "inbound",
    "network_ip": "127.0.0.1",
    "failure_status": "480",
    "failure_phrase": "Temporarily not available",
    "from_number": "+917123451234",
    "to_number": "1234567890",
    "hangup_cause": "NORMAL_CLEARING",
    "sip_fields": {
      "livekit_id": "c12ac441-399c-45d8-bc99-ae187cc3bfcc"
    },
    "timestamp": "2026-02-18T11:59:42.466362Z"
  }
}

Sample Call Quality Event


Code
{
    "event_name": "iravoice::call_quality",
    "event_data": {
        "tenant_id": "xyz",
        "call_uuid": "62b0ab5e-483d-4e61-8143-ecde97e36d23",
        "call_params": {
            "call_direction": "outbound"
        },
        "dialer_ip": "192.168.11.13",
        "gateway": "",
        "call_type": "proxy",
        "direction": "inbound",
        "network_ip": "127.0.0.1",
        "mos": "4.50",
        "quality": "100.00",
        "sip_fields": {
          "livekit_id": "c12ac441-399c-45d8-bc99-ae187cc3bfcc"
        },
        "timestamp": "2026-02-18T11:59:42.765432Z"
    }
}

Support

For SIP credentials, routing configuration, recording setup, webhook integration, or troubleshooting assistance, please contact the Epicode Support Team.

Last modified on June 30, 2026

Server Monitor Vapi