# 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 ![](/docs/IraVoice-livekit.png) ### Outbound Call Flow B[IraVoice] B --> C[PSTN Provider] C --> D[Phone User] `} /> ### Inbound Call Flow B[PSTN Provider] B --> C[IraVoice] C --> D[LiveKit Room] `} /> ## 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. :::note **Reference:** https://docs.livekit.io/telephony/making-calls/outbound-trunk/ ::: #### Example ```json { "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) :::note **Reference:** https://docs.livekit.io/telephony/making-calls/outbound-calls/ ::: :::note ### 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 ```json { "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 ```text {destination_number}@abcdefghijk.sip.livekit.cloud;transport=tcp ``` #### Example ```text +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. :::note **Reference:** https://docs.livekit.io/telephony/accepting-calls/inbound-trunk/ ::: #### Example ```json { "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. :::note **Reference:** https://docs.livekit.io/telephony/accepting-calls/inbound-trunk/ ::: #### Example ```json { "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: 1. Push recordings to a customer-provided HTTP endpoint (webhook). 2. 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 ```json { "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 ```json { "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 ```json { "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 ```json { "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" } } ``` --- :::note ### Support For SIP credentials, routing configuration, recording setup, webhook integration, or troubleshooting assistance, please contact the **Epicode Support Team**. :::