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 --domain=meet.example.com
After Jigasi is started it will register to the XMPP server and connect to the brewery room.
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).
- --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 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. 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.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 SIP dial button and
using 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. 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:
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
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 properly in ~/jigasi/jigasi-home/sip-communicator.properties:
org.jitsi.jigasi.transcription.customService=org.jitsi.jigasi.transcription.VoskTranscriptionService
Finally, configure the websocket URL of the VOSK service in ~/jigasi/jigasi-home/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"}
To use LibreTranslate
for translation, configure the following properties in ~/jigasi/jigasi-home/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 ~/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.
| 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 or not to advertise the URL which will serve the final transcript when Jigasi joins the room. |
| org.jitsi.jigasi.transcription.SAVE_JSON | false | Whether or not to save the final transcript in JSON. Note that this format is not very human readable. |
| org.jitsi.jigasi.transcription.SAVE_TXT | true | Whether or not to save the final transcript in plain text. |
| org.jitsi.jigasi.transcription.SEND_JSON | true | Whether or not 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 or not to send results, when they come in, to the chatroom in plain text. Note that this will result in the chat being somewhat spammed. |
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.