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https://github.com/jitsi/jigasi

Jigasi: a server-side application acting as a gateway to Jitsi Meet conferences. Currently allows regular SIP clients to join meetings and provides transcription capabilities.
https://github.com/jitsi/jigasi

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Jigasi: a server-side application acting as a gateway to Jitsi Meet conferences. Currently allows regular SIP clients to join meetings and provides transcription capabilities.

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README

        

Jigasi
======

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

Install and run
============

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

[quick install instructions]: https://github.com/jitsi/jitsi-meet/blob/master/doc/quick-install.md

1. Checkout latest source:

```
git clone https://github.com/jitsi/jigasi.git
```
2. Build:

```
cd jigasi
mvn install -Dassembly.skipAssembly=false
```

3. Extract - choose either `jigasi-linux-x64-{version}.zip`, `jigasi-linux-x86-{version}.zip` or `jigasi-macosx-{version}.zip` based on the system.

```
cd target/
unzip jigasi-{os-version}-{version}.zip
```

4. Configure a muc component in your XMPP server that will be used for the brewery rooms. If your server is Prosody: edit /etc/prosody/prosody.cfg.lua or the appropriate file in /etc/prosody/conf.d and append following lines to your config (assuming that domain 'meet.example.com'):

```
Component "internal.auth.meet.example.com" "muc"
storage = "memory"
modules_enabled = {
"ping";
}
admins = { "[email protected]", "[email protected]" }
muc_room_locking = false
muc_room_default_public_jids = true
```

5. Setup SIP account

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

6. Setup the xmpp account for jigasi control room (brewery).
prosodyctl register jigasi auth.meet.example.com topsecret
Replace ```<>``` tag with Base64 encoded password (topsecret) in the sip-communicator.properties file.

6. Start Jigasi

```
cd jigasi/target/jigasi-{os-version}-{version}/
./jigasi.sh
```
After Jigasi is started it will register to the XMPP server and connect to the brewery room.

How it works
============

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 signaling, ICE, DTLS/SRTP termination and multiple-SSRC handling for them.

Outgoing calls
==============

To call someone from Jitsi Meet application, Jigasi must be configured and started like described in the 'Install and run' section. From the invite dialog from the Participants pane you can invite (dial-out) telephone participants.

Incoming calls
==============

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': '[email protected]"' will cause Jigasi to join the conference 'https://meet.example.com/room1234' (assuming that our domain is 'meet.example.com').

Configuring SIP and Transcription
=======================================

It is possible to either enable or disable the functionality of SIP and
transcription. By default, the properties
`org.jitsi.jigasi.ENABLE_TRANSCRIPTION=false`
and
`org.jitsi.jigasi.ENABLE_SIP=true`
in
`jigasi-home/sip-communicator.properties`
enable SIP and disable transcription. To change this, simple set the desired
property to `true` or `false`.

Using Jigasi to transcribe a Jitsi Meet conference
==================================================

It is also possible to use Jigasi as a provider of nearly real-time transcription
as well as translation while a conference is ongoing as well as serving a complete transcription
after the conference is over. This can be done by using the `Subtitles` button from the menu in jitsi-meet.

Currently, Jigasi can send speech-to-text results to jitsi-meet as either plain text or JSON. If it's send as JSON,
Jitsi Meet will provide subtitles in 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 if that is configured.

To configure jigasi as a transcriber in a meeting, you will need to have it log
in with a specific domain that is set as hidden in jitsi-meet config.
In prosody config (/etc/prosody/conf.d/meet.example.com.cfg.lua) you need to have:

```
VirtualHost "recorder.meet.example.com"
modules_enabled = {
"ping";
}
authentication = "internal_hashed"
```

Restart prosody if you have added the virtual host config and then create the transcriber account:

```
prosodyctl register transcriber recorder.yourdomain.com jigasirecorderexamplepass
```

Edit the `/etc/jitsi/meet/meet.example.com-config.js` file, add/set the following:

```
config.hiddenDomain = "recorder.meet.example.com";
config.transcription = { enabled: true };
```

And in jigasi config (`/etc/jitsi/jigasi/sip-communicator.properties`):

```
org.jitsi.jigasi.ENABLE_SIP=false
org.jitsi.jigasi.ENABLE_TRANSCRIPTION=true
org.jitsi.jigasi.xmpp.acc.USER_ID=transcriber@recorder.meet.example.com
org.jitsi.jigasi.xmpp.acc.PASS=jigasirecorderexamplepass
org.jitsi.jigasi.xmpp.acc.ANONYMOUS_AUTH=false
org.jitsi.jigasi.xmpp.acc.ALLOW_NON_SECURE=true
```

Configure a transcription provider(Google, Vosk etc.) and restart jigasi.

Jigasi supports multiple transcription services, including Google Cloud speech-to-text
API, Vosk speech recognition server, a custom flavor of Whisper
and Oracle Cloud AI Speech.

Google configuration for transcription
====================

For Jigasi to act as a transcriber, it sends the audio of all participants in the
room to an external speech-to-text service. To use [Google Cloud speech-to-text API](https://cloud.google.com/speech/)
it is required to install the [Google Cloud SDK](https://cloud.google.com/sdk/docs/)
on the machine running Jigasi. To install on a regular debian/ubuntu environment:

```
curl https://packages.cloud.google.com/apt/doc/apt-key.gpg | sudo gpg --dearmor -o /usr/share/keyrings/cloud.google.gpg
echo "deb [signed-by=/usr/share/keyrings/cloud.google.gpg] https://packages.cloud.google.com/apt cloud-sdk main" | sudo tee -a /etc/apt/sources.list.d/google-cloud-sdk.list
sudo apt-get update && sudo apt-get install google-cloud-cli

gcloud init
gcloud auth application-default login
```

You will generate a file used for authentication of Google cloud api in Jigasi. You will see a result like:
`Credentials saved to file: [/root/.config/gcloud/application_default_credentials.json]`

Move the file to Jigasi config and change its permissions:

```
mv /root/.config/gcloud/application_default_credentials.json /etc/jitsi/jigasi
chown jigasi:jitsi /etc/jitsi/jigasi/application_default_credentials.json
```

In the file `/etc/jitsi/jigasi/config` add at the end:

```
# Credential for Google Cloud Speech API
GOOGLE_APPLICATION_CREDENTIALS=/etc/jitsi/jigasi/application_default_credentials.json
```

and restart Jigasi.

Vosk configuration for transcription
==================

To use [Vosk speech recognition server](https://github.com/alphacep/vosk-server)
start the server with a docker:

```
docker run -d -p 2700:2700 alphacep/kaldi-en:latest
```

Then configure the transcription class with the following property in `/etc/jitsi/jigasi/sip-communicator.properties`:

```
org.jitsi.jigasi.transcription.customService=org.jitsi.jigasi.transcription.VoskTranscriptionService
```

Finally, configure the websocket URL of the VOSK service in `/etc/jitsi/jigasi/sip-communicator.properties`:

If you only have one instance of VOSK server:

```
org.jitsi.jigasi.transcription.vosk.websocket_url=ws://localhost:2700
```

If you have multiple instances of VOSK for transcribing different languages, configure
the URLs of different VOSK instances in JSON format:
```
# org.jitsi.jigasi.transcription.vosk.websocket_url={"en": "ws://localhost:2700", "fr": "ws://localhost:2710"}
```

Whisper configuration for transcription
==================

If you plan to use our own flavor of Whisper (check [jitsi/skynet](https://github.com/jitsi/skynet)), start by
configuring the following properties in `/etc/jitsi/jigasi/sip-communicator.properties`:

```
org.jitsi.jigasi.transcription.customService=org.jitsi.jigasi.transcription.WhisperTranscriptionService
org.jitsi.jigasi.transcription.whisper.websocket_url=wss://:<>
```

If you also plan to enable the ASAP authentication, have a look at the
[documentation](https://github.com/jitsi/skynet/blob/master/docs/streaming_whisper_module.md) and at the properties
in the transcription options section of this README.

Oracle Cloud AI Speech configuration for transcription
==================

To use [Oracle Cloud AI Speech](https://docs.oracle.com/en-us/iaas/Content/speech/home.htm), you need to configure the
following properties in `/etc/jitsi/jigasi/sip-communicator.properties`:

```
org.jitsi.jigasi.transcription.customService=org.jitsi.jigasi.transcription.OracleTranscriptionService
org.jitsi.jigasi.transcription.oci.websocketUrl=wss://realtime.aiservice-<>.<>.oci.oraclecloud.com
```

You also need to place valid OCI credentials under `/usr/share/jigasi/.oci`. Or point to a different location by setting
the `OCI_CONFIG_FILE` environment variable.

LibreTranslate configuration for translation
==================

To use [LibreTranslate](https://github.com/LibreTranslate/LibreTranslate)
for translation, configure the following properties in `/etc/jitsi/jigasi/sip-communicator.properties`:

```
org.jitsi.jigasi.transcription.translationService=org.jitsi.jigasi.transcription.LibreTranslateTranslationService
org.jitsi.jigasi.transcription.libreTranslate.api_url=http://localhost:5000/translate
```

Run the docker container along with Jigasi:
```
docker run -d -p 5000:5000 libretranslate/libretranslate
```
Note that by default, the LibreTranslate server downloads all language models
before starting to listen to requests. You may refer to the
[documentation](https://github.com/LibreTranslate/LibreTranslate/blob/main/README.md)
to set up a volume or set the available languages to reduce download time.

Transcription options
=====================

There are several configuration options regarding transcription. These should
be placed in `/etc/jitsi/jigasi/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.


Property name
Default value
Description


org.jitsi.jigasi.transcription.DIRECTORY
/var/lib/jigasi/transcripts
The folder which will be used to store and serve the final
transcripts.


org.jitsi.jigasi.transcription.BASE_URL
http://localhost/
The base URL which will be used to serve the final transcripts.
The URL used to serve a transcript will be this base appended by the
filename of the transcript.


org.jitsi.jigasi.transcription.jetty.port
-1
The port which will be used to serve the final transcripts. Its
default value is -1, which means the Jetty instance serving the
transcript files is turned off.


org.jitsi.jigasi.transcription.ADVERTISE_URL
false
Whether to advertise the URL which will serve the final
transcript when Jigasi joins the room.


org.jitsi.jigasi.transcription.SAVE_JSON
false
Whether to save the final transcript in JSON. Note that this
format is not very human-readable.


org.jitsi.jigasi.transcription.SAVE_TXT
true
Whether to save the final transcript in plain text.


org.jitsi.jigasi.transcription.SEND_JSON
true
Whether to send results, when they come in, to the chatroom
in JSON. Note that this will result in subtitles being shown.


org.jitsi.jigasi.transcription.SEND_TXT
false
Whether to send results, when they come in, to the chatroom
in plain text. Note that this will result in the chat being somewhat
spammed.


org.jitsi.jigasi.transcription.remoteTranscriptionConfigUrl


Makes a GET request to https://YOUR-URL/tenant in order to retrieve which transcription service to use.
It expects a JSON response with the transcriberType key set to one of the following values:
GOOGLE, EGHT_WHISPER (see jitsi/skynet),
ORACLE_CLOUD_AI_SPEECH or VOSK. If the response is invalid or the request fails,
it will try to use the value of org.jitsi.jigasi.transcription.customService. If no value is
set, it will not make the request.



org.jitsi.jigasi.transcription.remoteTranscriptionConfigUrl.key

Base64 RSA256 private key to sign an ASAP JWT with when issuing the request above.


org.jitsi.jigasi.transcription.remoteTranscriptionConfigUrl.kid

The key's ID.


org.jitsi.jigasi.transcription.remoteTranscriptionConfigUrl.aud

The JWT audience.


org.jitsi.jigasi.transcription.customService


Which transcription service to use between GoogleCloudTranscriptionService, WhisperTranscriptionService
(see jitsi/skynet), OracleTranscriptionService and
VoskTranscriptionService.



org.jitsi.jigasi.transcription.google_model
latest_long

The model used by the Google speech-to-text API, check the available models
here.



org.jitsi.jigasi.transcription.whisper.private_key


A base64 RSA256 private key to sign an ASAP JWT with when
org.jitsi.jigasi.transcription.WhisperTranscriptionService is chosen.



org.jitsi.jigasi.transcription.whisper.private_key_name

The key ID for the org.jitsi.jigasi.transcription.WhisperTranscriptionService JWT.


org.jitsi.jigasi.transcription.whisper.jwt_audience

The audience for the org.jitsi.jigasi.transcription.WhisperTranscriptionService JWT.


org.jitsi.jigasi.transcription.whisper.websocket_url
ws://localhost:8000/ws

The websocket URL for the org.jitsi.jigasi.transcription.WhisperTranscriptionService
transcription service.



org.jitsi.jigasi.transcription.oci.websocketUrl


The websocket url for the org.jitsi.jigasi.transcription.OracleTranscriptionService
transcription service.



org.jitsi.jigasi.transcription.oci.compartmentId


The compartment ID for the org.jitsi.jigasi.transcription.OracleTranscriptionService
transcription service.

Call control MUCs (brewery)
=======================================

For outgoing calls jigasi by default configures using a control room called brewery(XMPP MUC).
To configure using MUCs you need to add an XMPP account that will be used to
connect to the XMPP server and add the property
`org.jitsi.jigasi.BREWERY_ENABLED=true`.
Here are example XMPP account properties:
```
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1=acc-xmpp-1
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.ACCOUNT_UID=Jabber:[email protected]
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.USER_ID=jigasi@auth.meet.example.com
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_SERVER_OVERRIDDEN=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.SERVER_ADDRESS=
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.SERVER_PORT=5222
#net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.BOSH_URL=https://xmpp_server_ip_address/http-bind
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.ALLOW_NON_SECURE=true
#base64 AES keyLength:256 or 128
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.PASSWORD=
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.AUTO_GENERATE_RESOURCE=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.RESOURCE_PRIORITY=30
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.KEEP_ALIVE_METHOD=XEP-0199
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.KEEP_ALIVE_INTERVAL=30
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.CALLING_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.JINGLE_NODES_ENABLED=false
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_CARBON_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.DEFAULT_ENCRYPTION=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_USE_ICE=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_ACCOUNT_DISABLED=false
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_PREFERRED_PROTOCOL=false
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.AUTO_DISCOVER_JINGLE_NODES=false
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.PROTOCOL=Jabber
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_USE_UPNP=false
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IM_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.SERVER_STORED_INFO_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_FILE_TRANSFER_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.USE_DEFAULT_STUN_SERVER=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.ENCRYPTION_PROTOCOL.DTLS-SRTP=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.ENCRYPTION_PROTOCOL_STATUS.DTLS-SRTP=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.OVERRIDE_ENCODINGS=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.G722/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.GSM/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.H263-1998/90000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.H264/90000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.PCMA/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.PCMU/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.SILK/12000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.SILK/16000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.SILK/24000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.SILK/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.VP8/90000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.iLBC/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.opus/48000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.speex/16000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.speex/32000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.speex/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.BOSH_URL_PATTERN=https://{host}{subdomain}/http-bind?room={roomName}
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.DOMAIN_BASE=meet.example.com
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.BREWERY=JigasiBrewery@internal.auth.meet.example.com

```
The property `BOSH_URL_PATTERN` is the bosh URL that will be used from jigasi
when a call on this account is received.

The value of `BREWERY` is the name of the brewery room where jigasi will connect.
That room needs to be configured in jicofo with the following property:
`org.jitsi.jicofo.jigasi.BREWERY=JigasiBrewery@internal.auth.meet.example.com` or in the new jicofo config:
`hocon -f /etc/jitsi/jicofo/jicofo.conf set jicofo.jigasi.brewery-jid '"[email protected]"'`
Where prosody needs to have a registered muc component: `internal.auth.meet.example.com`.

The configuration for the XMPP control MUCs that jigasi uses can be modified at
run time using REST calls to `/configure/`.

# Adding an XMPP control MUC
A new XMPP control MUC can be added by posting a JSON which contains its configuration to `/configure/call-control-muc/add`:
```
{
"id": "acc-xmpp-1",
"ACCOUNT_UID":"Jabber:[email protected]@meet.example.com",
"USER_ID":"[email protected]",
"IS_SERVER_OVERRIDDEN":"true",
.....
}
```

If a configuration with the specified ID already exists, the request will
succeed (return 200), but the configuration will NOT be updated. If you need to
update an existing configuration, you need to remove it first and then re-add it.

# Removing an XMPP control MUC.
An XMPP control MUC can be removed by posting a JSON which contains its ID
to `/configure/call-control-muc/remove`:
```
{
"id": "acc-xmpp-1"
}
```

The request will be successful (return 200) as long as the format of the JSON is
as expected, and the connection was found and removed.