Configuring Trunk and Dispatch on self-hosted SIP Server

[osyris]

I think I am missing something obvious. I've created a self-hosted SIP server that is run natively without Docker. I want to use it with LiveKit Cloud Server. My SIP Sever config.yaml is:

log_level: debug
api_key: xxxxxxx
api_secret: xxxxxxx
ws_url: wss://xxxxxx-5cb9cj27.livekit.cloud
health_port: 8080
sip_port: 5060
rtp_port: 10000-20000
use_external_ip: true
redis:
  address: localhost:6379

Server is starting and running nicely.

In the Docs there are instructions on how to add Inbound Trunk and Dispatch Rule but those are added to the LiveKit Cloud.

Is this part done correctly? When using self-hosted SIP Server, do I still need to add Trunk and Dispatch config to the LiveKit Cloud or does need to go somewhere into the self-hosted SIP Server config?

On the Twilio side I also have the SIP-trunk configured and when I call my number I see in the SIP Server logs:

Jan 09 21:39:01 4066261-fk23777 sip[3081]: 2025-01-09T21:39:01.387+0300        INFO        sip        sip/inbound.go:173        processing invite        {"nodeID": "NE_7PbsSX7sWPnH", "callID": "SCL_ZxHsyRYehWg6", "fromIP": "54.171.127.194", "toIP": "188.225.72.77:5060", "fromHost": "rodion-trunk.pstn.twilio.com", "fromUser": "+3725258198", "toHost": "188.225.72.77", "toUser": "+3726510312", "sipTag": "75348722_c3356d0b_11fbc320-60be-4765-b8b7-5a1734195fe9", "sipCallID": "c72948d3b463d8ff6f1e5e5556812462@0.0.0.0", "twilioCallSID": "CA3cb523806805f311638778f6ad3a63d1", "twilioAccSID": "AC75f32eb24a42a26e29ac1a086798f125"}
Jan 09 21:39:02 4066261-fk23777 sip[3081]: 2025-01-09T21:39:02.388+0300        WARN        sip        sip/inbound.go:209        Rejecting inbound, auth check failed        {"nodeID": "NE_7PbsSX7sWPnH", "callID": "SCL_ZxHsyRYehWg6", "fromIP": "54.171.127.194", "toIP": "188.225.72.77:5060", "fromHost": "rodion-trunk.pstn.twilio.com", "fromUser": "+3725258198", "toHost": "188.225.72.77", "toUser": "+3726510312", "sipTag": "75348722_c3356d0b_11fbc320-60be-4765-b8b7-5a1734195fe9", "sipCallID": "c72948d3b463d8ff6f1e5e5556812462@0.0.0.0", "twilioCallSID": "CA3cb523806805f311638778f6ad3a63d1", "twilioAccSID": "AC75f32eb24a42a26e29ac1a086798f125", "error": "no response from servers"}

It says "no response from servers", but what servers is it not getting the response from?

In Twilio the Origination URL is:

SID                                 Friendly Name  Sip URL            Priority  Weight
OU4923984c171009f70f8de239b496218a  null           sip:188.225.72.77  1         1   

Does my config look correct? What am I missing to make this work?

[dennwc]

This is not a supported configuration, I'm afraid. If you want to self-host, you must run both LiveKit server and LiveKit SIP.

The reason why it currently fails is that you host a separate Redis instance. It is used by LiveKit services as a shared RPC bus. Thus, your SIP server cannot reach LiveKit APIs on the cloud, because it's connecting to a local Redis, not the Cloud RPC bus.

[osyris]

Thanks for confirming this! After reading the docs further I also found the description of how Redis is used to as the connection layer so it became obvious that my initial setup will not work :)

I am already running both LK Server and LK SIP and everything is working as expected 👌

[devb]

@osyris @dennwc - I have also installed the LK server and LK sip on my instance.. however, can you guide me on using the lk sip apis using cli on my instance?? As LK server and LK sip server are running inside docker.. where should I run these commands to create inbound/outbound trunk and dispatch rules?? on VM level OR inside the docker container? Please help
Thanks in advance.

[rajshahunify]

@devb if lk server and lk sip are on the same server ,

You do not need to enter the Docker containers. The lk CLI communicates with your containers over the network (localhost).

How to do it:

Install the CLI on your VM:

Bash

curl -sSL https://get.livekit.io/cli | bash
Set your credentials (on the VM terminal): Use the same API Key/Secret defined in your livekit.yaml or docker-compose.yml.

Bash

export LIVEKIT_URL=http://localhost:7880
export LIVEKIT_API_KEY=your_api_key
export LIVEKIT_API_SECRET=your_api_secret
Run the command:

Bash

lk sip inbound create <args...>

examples::

Yes, here are the exact commands.

Since the CLI requires a JSON configuration file as input, you cannot do it in just one line. You must first create a small file defining the rules, then run the command to apply it.

1. Create an "Inbound Trunk"
   This tells LiveKit: "Accept calls for this phone number."

A. Create the file inbound.json: (Replace +15550001234 with the number FreeSWITCH is sending to LiveKit. If you are just testing, you can often use default or the extension number like 1000).

JSON

{
"trunk": {
"name": "freeswitch-trunk",
"numbers": ["+15550001234"],
"krisp_enabled": false
}
}
B. Run the command:

Bash

lk sip inbound create inbound.json
Note the SIPTrunkID in the output (e.g., ST_xxxxxx). You will need it if you want to be specific in the next step, though usually it's not strictly required if you have only one trunk.

2. Create a "Dispatch Rule"
   This tells LiveKit: "When a call comes in, send it to this Room."

A. Create the file dispatch.json:

JSON

{
"name": "General Routing",
"rule": {
"dispatchRuleDirect": {
"roomName": "my-voice-room",
"pin": "1234"
}
}
}
(Note: pin is optional. If you remove it, anyone can join).

B. Run the command:

Bash

lk sip dispatch create dispatch.json
3. (Optional) Create an "Outbound Trunk"
Only do this if you want LiveKit to be able to dial out to FreeSWITCH (to make calls to real phones).

A. Create the file outbound.json:

JSON

{
"trunk": {
"name": "freeswitch-outbound",
"address": "172.17.0.1",
"numbers": ["+15550001234"],
"auth_username": "livekit_user",
"auth_password": "your_password"
}
}
address: This should be your FreeSWITCH IP. If running in Docker, 172.17.0.1 is often the host IP.

auth: Credentials you set up in FreeSWITCH for this extension.

B. Run the command:

Bash

lk sip outbound create outbound.json
Summary of Workflow
Create JSON file (e.g., nano inbound.json -> paste -> save).

Run Command (lk sip inbound create inbound.json).

Done. LiveKit now knows how to handle the traffic.

hope this helps

[naaevi]

Thanks @rajshahunify for your reply, but I've figured it out long back. 👍