Jitsi Meet Jigasi Server

Overview

The jitsimeet jigasi server provides integration with a SIP gateway enabling call-in and dial-out features for Jitsi Meet and also transcription services via a variety of back ends.

We are using the transcription service via the Jitsi meet skynet API which uses the Faster Whisper LLM models, which are based on the OpenAI Whisper models, which is a general purpose speech recognition model. This LLM is run on May First infrastructure and does not leak any conversations.

We have the option of using the dial-in service with a phone number provided via the corporate Twilio service. However, any Jitsi meet conference with a dial-in participant is not encrypted and can be leaked.

Jigasi can be installed on the same server running Jitsi meet. But, the skynet API should be installed on a dedicated server.

Setting up Jigasi with transcription service

When setup with transcription, the Jigasi service, which runs on the same server as jitsimeet, connects to a separate server running skynet.

skynet is the Jitsi Meet generic API for providing LLM services. And, at the moment, it uses Faster Whisper, which is a faster variant of the Open AI Whisper model for converting audio to text. No data is shared outside our network. The details are in the ansible role.

Setting up Jigasi with a call-in Number

We do not have a call-in number configured. This is how it would be setup.

Here are the general steps that happen when a jitsi meet meeting is setup with support for a call in phone number.

  1. The meeting is created by the meeting organizer via Jitsi Meet by simply starting a meeting.
  2. The organizer clicks the participants icon and then clicks the button to invite more participants.
  3. This triggers two API calls:
    • One call to the web service API that provides a list of dial in numbers that this Jitsi Meet instance supports (configured using the dialInNumbersUrl Jitsi Meet setting in /etc/jitsi/meet/domain-config.js file on the jisti meet server)
    • One call to the server serivce API (configured using the dialInConfCodeUrl setting) that translates a conference name for a 6 digit number that a phone caller will enter into their key pad to be directed to the right conference. If this is the first time the API has seen a given conference name, then it generates a random 6 digit number and stores that number along with the conference name in a database for future lookups.
  4. Based on the results of both API calls, the Jitsi Meet interface displays the link to follow to join the meeting plus the call in numbers and the code to be used for people who want to join via phone.
  5. The organizer copies and pastes this info into a text message, email message, etc to notify participants of how to join the call.
  6. If a participant dials the phone number, Twilio prompts the caller to enter the 6 digit pin and then sends that PIN to our API service.
  7. Our API service looks up the corresponding conference room for that pin and sends an XML response back to Twilio directing Twilio to transfer the caller to our Jibri instance with the SIP header X-Room-Name set to the proper conference room.
  8. Our Jibri instance connects that incoming SIP caller to the running conference.

Getting a Twilio phone number

The first step in setting up jigasi is to get a Twilio phone number. The Twilio interface may have changed, but as of May 2021, the process was:

  1. Go to the Phone Numbers menu in Twilio
  2. Click Buy a Number
  3. Don’t configure the number (yet)
  4. Click Programmable Voice -> SIP domains
  5. Click the “+” sign to add a new SIP domain.
  6. Add a Friendly Name and SIP URI
  7. Under Voice Authentication, click the “+” sign next to Credential List and create a username and password (use your phone number as the username, without the 1).
  8. Under Call Control Configuration:
    • Configure with: Webhooks, TwiML Bins, Functions, Studio, Proxy
    • A Call Comes In: TwiML Bin, then click “+” to add a TwiML Bin with the content (replace phone_number and domain): 
       <Response> 
   <Dial answerOnBridge="true"> 
     <Sip> sip:{{ phone_number }}@{{ domain }}.sip.us1.twilio.com </Sip> 
   </Dial> 
 </Response>
  9. Under SIP Registration, click to Enable
  10. Under SIP Registration Authentication, choose the same Credentials List you created above
  11. Save the SIP domain.
  12. Go back to the Numbers screen and open your phone number
  13. Go to the “A Call Comes in” section a. Choose “TwiML Bin” b. Select the same TwiML Bin you created above.

Variables

In the server’s yaml file, define the twilio sip login (e.g. 1234567890@orgname.sip.us1.twilio.com) as jigasi_sip_login.

Record the twilio password in the ansible secrets.txt vaule with the key: vault_jibri_sip_password

You will also need the following:

  • jigasi_jitsimeet_conference_mapper_git_repo (set to: https://code.mayfirst.org/mfmt/jitsimeet-conference-mapper.git)
  • jigasi_jitsimeet_conference_mapper_version (set to the latest commit)
  • jigasi_dialin_numbers (set to your Twilio number)

The post install manual step.

Wah. This is so annoying. The XMPP password is defined by the installer, but frustratingly not provided in jigasi/sip-communicator.properties. To make matters more ridiculous, it’s defined in this file base64 encoded, but in the spot we need it, it has to be plain text. So there’s no way we can complete this automatically. Sheesh.

Here are the manual instructions:

  1. Edit the file: /etc/jitsi/jigasi/sip-communicator.properties
  2. Find the setting: net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PASSWORD
  3. Echo the value to the base64 -d command to decode the password
  4. Find the entry: org.jitsi.jigasi.xmpp.acc.PASS=
  5. Uncomment it and set it to the decoded value of the XMPP password.

The rest

The rest of the ansible role is dedicated to setting up our Jitsi Meet Conference mapper API. This project is managed in it’s own git repo with Nginx configuration, systemd start up and letsencrypt support.