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.
- Checkout latest source:
git clone https://github.com/jitsi/jigasi.git
- Build:
cd jigasi
mvn install -Dassembly.skipAssembly=false
- Extract - choose either
jigasi-linux-x64-{version}.zip,jigasi-linux-x86-{version}.ziporjigasi-macosx-{version}.zipbased on the system.
cd target/
unzip jigasi-{os-version}-{version}.zip
- 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 = { "focus@auth.meet.example.com", "jigasi@auth.meet.example.com" }
muc_room_locking = false
muc_room_default_public_jids = true
- Setup SIP account
Go to jigasi/jigasi-home and edit sip-communicator.properties file. Replace <<JIGASI_SIPUSER>> tag with SIP username for example: "user1232@sipserver.net". Then put Base64 encoded password in place of <<JIGASI_SIPPWD>>.
-
Setup the xmpp account for jigasi control room (brewery). prosodyctl register jigasi auth.meet.example.com topsecret Replace
<<JIGASI_XMPP_PASSWORD_BASE64>>tag with Base64 encoded password (topsecret) in the sip-communicator.properties file. -
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.
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.
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.
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@conference.meet.example.com"' will cause Jigasi to join the conference 'https://meet.example.com/room1234' (assuming that our domain is 'meet.example.com').
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.
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.
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 it is required to install the Google Cloud SDK 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.
To use Vosk speech recognition 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"}
If you plan to use our own flavor of Whisper (check 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://<YOUR-DOMAIN>:<<PORT>>
If you also plan to enable the ASAP authentication, have a look at the documentation and at the properties in the transcription options section of this README.
To use Oracle Cloud AI Speech, 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-<<ENV>>.<<REGION>>.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.
To use 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 to set up a volume or set the available languages to reduce download time.
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.
|
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:jigasi@auth.meet.example.com
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=<xmpp_server_ip_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=<xmpp_account_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 '"JigasiBrewery@internal.auth.meet.example.com"'
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/.
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:jigasi@auth.meet.example.com@meet.example.com",
"USER_ID":"jigasi@auth.meet.example.com",
"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.
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.