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Purpose of this document

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Configuration in Telnyx Mission Control Portal

• 1. Have an account created on Telnyx portal
• 2. Complete L2 verification process
• 3. Purchase a number to be used for the voice calls

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1. SIP Connection

• In the Telnyx Mission Control Portal choose a “Real-Time Communications” -> “Voice” -> “SIP Trunking” menu on the left sidebar
• Click “Add SIP Connection” button

• Provide a name for your new SIP Connection
• Select FQDN as a type of connection
• Create a sip connection

• Choose “FQDN” as a connection type
• Provide SIP URI obtained from your LiveKit account. You can find it in your project settings page.
• Save FQDN settings with “Add” button

• Switch to the “Outbound Calls Authentication” tab in the Authentication & Routing Configuration section
• Choose “Credentials” for the Authentication Method
• Provide your user name and password which will be used later for the LiveKit outbound trunk configuration

• On top of SIP Connection page switch to the Outbound tab
• Select an existing Outbound Voice Profile from a dropdown list.
• If no profile is available you can create a one with a “Create New Outbound Voice Profile” link

• Finally save your sip connection with a “Save” button at the bottom of the page

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2. Programmable Voice Applications

• Call recording
• IVR (Interactive Voice Response)
• TTS (Text-To-Speech)
• STT (Speech-To-Text / Call transcription)
• Conversational AI (Voice bots with different AI models)
• and many more

• Switch to a “Real-Time Communications” -> “Voice” -> “Programmable Voice” menu on the left sidebar
• Create a new application with an “Add New App” button
• Provide a name for your application
• Configure webhooks URL to receive webhooks about call status changes

• In the Inbound section provide a subdomain name (later you need to use it in a LiveKit outbound trunk setup)
• In the Outbound section select an outbound voice profile
• Save the voice app configuration

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3. Outbound Voice Profile

• Switch to a “Real-Time Communications” -> “Voice” -> “Outbound Voice Profile” menu on the left sidebar
• Add a new Profile and provide a name for it
• Or you can adjust the existing profile with clicking on the edit icon

• Select the destinations which should be allowed to make outbound calls to
• Click “Save” button to confirm your configuration

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4. Number configuration

• Switch to a “Real-Time Communications” -> “Voice” -> “My Numbers” menu on the left sidebar
• For the number(s) you would like to use for the LiveKit voice participants, select your LiveKit sip connection from a dropdown list presented in the “SIP connections” column
• Update your configuration with a “Save” button
• You can purchase additional numbers in a “Buy Numbers” menu
• The same SIP connection may be assigned to multiple numbers

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Configuration in LiveKit

• 1. SIP URI is available in your project settings
• 2. LiveKit CLI is installed on your computer
• 3. ENV variables have been configured

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1. Create Inbound SIP Trunk

• Create a json file (inboundTrunk.json) with the configuration of your inbound sip trunk
• Provide a name for your trunk and a number to be used (or an array of the comma delimited numbers)

{
   "trunk": {
       "name": "Demo Inbound Trunk",
       "numbers": ["1234567890"]
   }
}

• Run LiveKit CLI command to create an inbound trunk

lk sip inbound create inboundTrunk.json

• In the response you will receive a trunk ID

SIPTrunkID: <your-trunk-id>

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2. Create Inbound Dispatch Rule

• Create a json file (dispatchRule.json)
• Provide a name for your rule, a trunk ID from the previous step and room name to which incoming calls should be connected

{
   "name": "Demo Dispatch Rule",
   "trunk_ids": ["<your-trunk-id>"],
   "rule": {
       "dispatchRuleDirect": {
           "roomName": "my-sip-room",
           "pin": ""
       }
   }
}

• Run LiveKit CLI command to create a dispatch rule

lk sip dispatch create dispatchRule.json

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3. Create Outbound SIP Trunk

• Create a json file (outboundTrunk.json) with the configuration of your outbound sip trunk
• Provide a name for your trunk
• In the address parameter provide “sip.telnyx.com” if you configured a SIP connection on Telnyx side
• If you want to make calls to the configured Voice App, for the address parameter provide a subdomain configured in the Inbound section of your voice application
• Provide username and password you set in your sip trunk configuration for the Outbound Calls Authentication.
• Credentials should be left blank when connecting to the subdomain of your Voice App.

{
   "trunk": {
       "name": "Demo Outbound Trunk",
       "address": "sip.telnyx.com",
       "numbers": ["1234567890"],
       "auth_username": "<outbound-user>",
       "auth_password": "<outbound-pass>"
   }
}

• Run LiveKit CLI command to create an outbound trunk

lk sip outbound create outboundTrunk.json

• In the response you will receive a trunk ID

SIPTrunkID: <your-trunk-id>

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4. Testing Outbound Calls

• To test an outbound call you can initiate a call with a CLI command
• Create a json file (sipParticipant.json) providing outbound trunk ID obtained in the previous step
• Provide a number to be called in a sip_call_to parameter
• Provide a room name, participant identity and name

{
	"sip_trunk_id": "<your-trunk-id>",
	"sip_call_to": "<phone-number-to-dial>",
	"room_name": "my-sip-room",
	"participant_identity": "sip-test",
	"participant_name": "Test Call"
}

• Run LiveKit CLI command to make an outbound call

lk sip participant create sipParticipant.json

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Troubleshooting

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1. Debugging tool

• Go to the “Debugging” menu under “Reporting” in the left menu
• Select “SIP Call Flow Tool” on the top bar
• Specify your search criteria and press “Search CDRs” button
• From the list of listed calls select a one with “Call Data Debugging” button

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References

• 1. Telnyx SIP trunk configuration data - https://sip.telnyx.com
• 2. Telnyx SIP trunk setup - https://developers.telnyx.com/docs/voice/sip-trunking/get-started
• 3. LiveKit SIP configuration - https://docs.livekit.io/sip/

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