Hello all.
I am trying to add transcription feature in jitsi meet application .My sip gateway is working properly.But confused from where to strat for working on transcription feature.Is there need of any external library?
Have you followed the instructions here? GitHub - 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.
Yes ,I followed this github link.But when I am trying to sign up in Google Cloud speech-to-text API it is asking my credit card details .As I am student cannot afford their cost for using their service.Ss,I want to ask that is there any other free source ?My sip audio calls are working smoothly .Just not clear about how to enable transcription.Could you please guide for moving further?
Thanks in advance.
You could look into using Vosk, check further in the readme. I think you should be able to qualify for some starter credit on Google Cloud though.
ok I will try .
Thank you for your valuable time .
Is there any reference available for configuring jigasi and domain.config file for working with Google Cloud speech to text api?As ,I am unable to find property which configure json file path of Google Cloud api ?If possible please guide me .
org.jitsi.jigasi.ENABLE_TRANSCRIPTION=true
org.jitsi.jigasi.ENABLE_SIP=true
whether to use the more expensive, but better performing
“video” model when doing transcription
org.jitsi.jigasi.transcription.USE_VIDEO_MODEL = false
org.jitsi.jigasi.transcription.GOOGLE_APPLICATION_CREDENTIALS=/etc/jitsi/jigasi/google.json
org.jitsi.jigasi.transcription.GOOGLE_CLOUD_PROJECT_ID=Id
org.jitsi.jigasi.transcription.ADAPTER_TYPE=google
delivering final transcript
org.jitsi.jigasi.transcription.DIRECTORY=/var/lib/jigasi/transcripts
org.jitsi.jigasi.transcription.DIRECTORY=/var/lib/jigasi/transcription
org.jitsi.jigasi.transcription.BASE_URL=http://localhost/
org.jitsi.jigasi.transcription.BASE_URL=http://localhost/
org.jitsi.jigasi.transcription.jetty.port=-1
org.jitsi.jigasi.transcription.jetty.port=-1
org.jitsi.jigasi.transcription.ADVERTISE_URL=false
save formats
org.jitsi.jigasi.transcription.SAVE_JSON=false
org.jitsi.jigasi.transcription.SAVE_TXT=true
org.jitsi.jigasi.transcription.SAVE_TXT=true
send formats
org.jitsi.jigasi.transcription.SEND_JSON=true
org.jitsi.jigasi.transcription.SEND_TXT=false
org.jitsi.jigasi.transcription.SEND_JSON=false
org.jitsi.jigasi.transcription.SEND_TXT=true
Configurations are like this.
==> jigasi.log <==
2023-05-26 08:45:19.272 INFO: [135] SipGatewaySession$SipCallStateListener.handleCallState#1448: [ctx=16850907127971291858232] Sip call IN_PROGRESS: Call: id=16850907127311532490330 peers=1
2023-05-26 08:45:19.274 INFO: [135] SipGatewaySession$SipCallStateListener.handleCallState#1450: [ctx=16850907127971291858232] SIP call format used: rtpmap:-1 opus/48000/2 fmtp:useinbandfec=1
2023-05-26 08:45:19.275 INFO: [135] SipGatewaySession$CallPeerListener.peerStateChanged#1521: [ctx=16850907127971291858232] SIP peer state: Connected
2023-05-26 08:45:19.278 INFO: [135] net.java.sip.communicator.service.protocol.media.CallPeerMediaHandler.start: Starting
2023-05-26 08:45:19.287 SEVERE: [137] net.java.sip.communicator.impl.protocol.sip.SipStackSharing.findTargetFor: no listeners
2023-05-26 08:45:19.287 SEVERE: [137] net.java.sip.communicator.impl.protocol.sip.SipStackSharing.processRequest: couldn’t find a ProtocolProviderServiceSipImpl to dispatch to
2023-05-26 08:45:19.328 INFO: [135] net.java.sip.communicator.service.protocol.media.TransportManager.sendHolePunchPacket: Send NAT hole punch packets
2023-05-26 08:45:19.357 SEVERE: [176] net.sf.fmj.media.Log.error: Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
2023-05-26 08:45:19.359 SEVERE: [176] net.sf.fmj.media.Log.error: Failed to prefetch: net.sf.fmj.media.ProcessEngine@31f9b5a
2023-05-26 08:45:19.376 SEVERE: [175] net.sf.fmj.media.Log.error: Error: Unable to prefetch net.sf.fmj.media.ProcessEngine@31f9b5a
2023-05-26 08:45:19.424 INFO: [109] JitsiOpenSslProvider.#52: jitsisrtp successfully loaded for OpenSSL 1.1
2023-05-26 08:45:19.802 SEVERE: [188] net.java.sip.communicator.impl.protocol.sip.SipStackSharing.findTargetFor: no listeners
2023-05-26 08:45:19.802 SEVERE: [188] net.java.sip.communicator.impl.protocol.sip.SipStackSharing.processRequest: couldn’t find a ProtocolProviderServiceSipImpl to dispatch to
2023-05-26 08:45:20.108 INFO: [109] Aes.benchmark#367: AES benchmark (of execution times expressed in nanoseconds): OpenSSL 1264, SunJCE 1447, BouncyCastle 19049 for AES/CTR/NoPadding
2023-05-26 08:45:20.109 INFO: [109] Aes.createCipher#433: Will employ AES implemented by OpenSSL for AES/CTR/NoPadding.
I am getting this jigasi logs
I am getting this subtitles icon but its not working .Do I need to do changes at backend or am I doing something wrong?
Could you please help me ?
I have created account on Google Cloud speech-to -text api.Also configured this in jigasi config file .
Still I am not getting any logs ,that will show my application is connected to Google Cloud speech-to-text api.
Also getting following errors:
net.java.sip.communicator.service.protocol.media.TransportManager.sendHolePunchPacket: Send NAT hole punch packets
2023-05-26 08:45:19.357 SEVERE: [176] net.sf.fmj.media.Log.error: Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
2023-05-26 08:45:19.359 SEVERE: [176] net.sf.fmj.media.Log.error: Failed to prefetch: net.sf.fmj.media.ProcessEngine@31f9b5a
2023-05-26 08:45:19.376 SEVERE: [175] net.sf.fmj.media.Log.error: Error: Unable to prefetch net.sf.fmj.media.ProcessEngine@31f9b5a
Are you on an arm64 machine by any chance?