Jitsi Gateway to SIP : a server-side application that links allows regular SIP clients to join Jitsi Meet conferences hosted by Jitsi Videobridge.

It is possible to install Jigasi along with Jitsi Meet using our quick install instructions or do this from sources using the instructions below.

1. Checkout latest source:

git clone https://github.com/jitsi/jigasi.git

2. Build:

cd jigasi
ant make

3. Configure external component in your XMPP server. If your server is Prosody: edit /etc/prosody/prosody.cfg.lua and append following lines to your config (assuming that subdomain is 'callcontrol' and domain 'meet.jit.si'):

Component "callcontrol.meet.jit.si"
    component_secret = "topsecret"

4. Setup SIP account

Go to jigasi/jigasi-home and edit sip-communicator.properties file. Replace <<JIGASI_SIPUSER>> tag with SIP username for example: "[email protected]". Then put Base64 encoded password in place of <<JIGASI_SIPPWD>>.

5. Start Jigasi

./jigasi.sh --domain=meet.jit.si --subdomain=callcontrol --secret=topsecret

After Jigasi is started it will register as XMPP component under the 'callcontrol' subdomain. In Jitsi Meet application -> config.js -> hosts.call_control must be set to 'callcontrol.meet.jit.si'. This will enable SIP calls in Jitsi Meet.

Supported arguments:

• --domain: specifies the XMPP domain to use.
• --host: the IP address or the name of the XMPP host to connect to (localhost by default).
• --port: the port of the XMPP host to connect on (5347 by default).
• --subdomain: subdomain name for SIP gateway component.
• --secret: the secret key for the sub-domain of the Jabber component implemented by this application with which it is to authenticate to the XMPP server to connect to.
• --min-port: the minimum port number that we'd like our RTP managers to bind upon.
• --max-port: the maximum port number that we'd like our RTP managers to bind upon.

Jigasi registers as a SIP client and can be called or be used by Jitsi Meet to make outgoing calls. Jigasi is NOT a SIP server. It is just a connector that allows SIP servers and B2BUAs to connect to Jitsi Meet. It handles the XMPP signalling, ICE, DTLS/SRTP termination and multiple-SSRC handling for them.

To call someone from Jitsi Meet application, Jigasi must be configured and started like described in the 'Install and run' section. This will cause the telephone icon to appear in the toolbar which will popup a call dialog on click.

Jigasi will register on your SIP server with some identity and it will accept calls. When Jigasi is called, it expects to find a 'Jitsi-Conference-Room' header in the invite with the name of the Jitsi Meet conference the call is to be patched through to. If no header is present it will join the room specified under 'org.jitsi.jigasi.DEFAULT_JVB_ROOM_NAME' config property. In order to change it, edit 'jigasi-home/sipcommunciator.properties' file.

Example:

Received SIP INVITE with room header 'Jitsi-Conference-Room': 'room1234' will cause Jigasi to join the conference 'https://meet.jit.si/room1234' (assuming that our domain is 'meet.jit.si').

It is also possible to use Jigasi as a provider of nearly real-time transcription while a conference is ongoing as well as serving a complete transcription after the conference is over. This can be done by using the SIP dial button and using the the URI jitsi_meet_transcribe. Currently Jigasi can send speech-to-text results to the chat of a Jitsi Meet room as either plain text or JSON. If it's send in JSON, Jitsi Meet will provide subtitles in the left corner of the video, while plain text will just be posted in the chat. Jigasi will also provide a link to where the final, complete transcript will be served when it enters the room.

For jigasi to act as a transcriber, it sends the audio of all participants in the room to an external speech-to-text service. Currently only the Google Cloud speech-to-text API is supported. It is required to install the Google Cloud SDK on the machine running Jigasi. To install on a regular debian/ubuntu environment:

export CLOUD_SDK_REPO="cloud-sdk-$(lsb_release -c -s)"
echo "deb http://packages.cloud.google.com/apt $CLOUD_SDK_REPO main" | sudo tee -a /etc/apt/sources.list.d/google-cloud-sdk.list
curl https://packages.cloud.google.com/apt/doc/apt-key.gpg | sudo apt-key add -
sudo apt-get update && sudo apt-get install google-cloud-sdk google-cloud-sdk-app-engine-java
gcloud init
gcloud auth application-default login

There are several configuration options regarding transcription. These should be placed in ~/jigasi/jigasi-home/sip-communicator.properties. The default value will be used when the property is not set in the property file. A valid XMPP account must also be set to make Jigasi be able to join a conference room.