make call to mobile with jigasi, there is big noise

[liangguohao66]

What happened?

jigasi ----> freeswitch ----VoiceGW ----mobile
freeswith got rtp like screenshot shows.

Platform

• Chrome (or Chromium based)
• Firefox
• Safari
• Other desktop browser
• Android browser
• iOS browser
• Electron app
• Android mobile app
• iOS mobile app
• Custom app using a mobile SDK

Browser / app / sdk version

14

Relevant log output

Reproducibility

• The problem is reproducible on meet.jit.si

More details?

No response

[damencho]

What is the codec that is used by jigasi on the sip side? Share your jigasi config masking any private information.

[liangguohao66]

What is the codec that is used by jigasi on the sip side? Share your jigasi config masking any private information.
PCMA is used after negotiation.

#Sample config with one XMPP and one SIP account configured
# Replace {sip-pass-hash} with SIP user password hash
# as well as other account properties

# Name of default JVB room that will be joined if no special header is included
# in SIP invite
#org.jitsi.jigasi.DEFAULT_JVB_ROOM_NAME=siptest@conference.<<DOMAIN_BASE>>

org.jitsi.jigasi.MUC_SERVICE_ADDRESS=conference.meeting.test.cn

net.java.sip.communicator.impl.protocol.SingleCallInProgressPolicy.enabled=false

# Should be enabled when using translator mode
#net.java.sip.communicator.impl.neomedia.audioSystem.audiosilence.captureDevice_list=["AudioSilenceCaptureDevice:noTransferData"]

# Adjust opus encoder complexity
net.java.sip.communicator.impl.neomedia.codec.audio.opus.encoder.COMPLEXITY=10

# Disables packet logging
net.java.sip.communicator.packetlogging.PACKET_LOGGING_ENABLED=false

# Enables support for startmuted, needs support in the IVR used for jigasi for users to be able to unmute
#org.jitsi.jigasi.ENABLE_SIP_STARTMUTED=false

# Enables support for visitors
#org.jitsi.jigasi.ENABLE_SIP_VISITORS=false

net.java.sip.communicator.impl.protocol.sip.acc1403273890647=acc1403273890647
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.ACCOUNT_UID=SIP\:1908@192.168.1.7
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PASSWORD=
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PROTOCOL_NAME=SIP
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.SERVER_ADDRESS=192.168.1.7
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PROXY_AUTO_CONFIG=false
#net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PROXY_ADDRESS=144.xx.x.xx
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PROXY_ADDRESS=192.168.1.7
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PROXY_PORT=50614
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PREFERRED_TRANSPORT=UDP
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.USER_ID=1908@192.168.1.7
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.KEEP_ALIVE_INTERVAL=25
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.KEEP_ALIVE_METHOD=OPTIONS
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.VOICEMAIL_ENABLED=false
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.AMR-WB/16000=750
#net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.G722/8000=700
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.G722/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.GSM/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.H263-1998/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.H264/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.PCMA/8000=600
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.PCMU/8000=650
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.SILK/12000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.SILK/16000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.SILK/24000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.SILK/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.VP8/90000=0
#net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.iLBC/8000=10
#net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.opus/48000=1000
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.red/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.speex/16000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.speex/32000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.speex/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.telephone-event/8000=1
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.ulpfec/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.OVERRIDE_ENCODINGS=true
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.DEFAULT_ENCRYPTION=false

# If an authenticated (hidden) domain is used to connect to a conference,
# PREVENT_AUTH_LOGIN will prevent the SIP participant from being seen as a
# hidden participant in the conference
#net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PREVENT_AUTH_LOGIN=FALSE

# Used when incoming calls are used in multidomain environment, used to detect subdomains
# used for constructing callResource and eventually contacting jicofo
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.DOMAIN_BASE=meeting.test.cn

# the pattern to be used as bosh url when using bosh in multidomain environment
#net.java.sip.communicator.impl.protocol.sip.acc1403273890647.BOSH_URL_PATTERN=https://{host}{subdomain}/http-bind?room={roomName}

# can be enabled to disable audio mixing and use translator, jigasi will act as jvb, just forward every ssrc stream it receives.
#net.java.sip.communicator.impl.protocol.sip.acc1403273890647.USE_TRANSLATOR_IN_CONFERENCE=true

# The following account is used only for dial-out from a meeting, it connects and join the brewery room used
# by Jicofo to discover available jigasi instances and use them (they can be used for outgoing calls or transcriptions if enabled or both)
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.meeting.test.cn
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.USER_ID=jigasi@auth.meeting.test.cn
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=127.0.0.1
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://127.0.0.1/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.RESOURCE=jigasi
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=false
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=false
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.BREWERY=jigasibrewery@internal.auth.meeting.test.cn
#net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.BOSH_URL_PATTERN=https://127.0.0.1:4431/http-bind?room={roomName}
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.BOSH_URL_PATTERN=https://meeting.test.cn:4431/http-bind?room={roomName}

net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.DOMAIN_BASE=meeting.test.cn

# we can receive dial/hangup only from the control muc
org.jitsi.jigasi.ALLOWED_JID=jigasibrewery@internal.auth.meeting.test.cn

# when checking other participants whether they are jibri/jigasi we can also check the the domain they use for connecting
#org.jitsi.jigasi.TRUSTED_DOMAINS=["recorder.<<DOMAIN_BASE>>", "sipjibri.<<DOMAIN_BASE>>", "jigasi.<<DOMAIN_BASE>>"]

org.jitsi.jigasi.BREWERY_ENABLED=true

# We can use the prefix org.jitsi.jigasi.xmpp.acc to override any of the
# properties that will be used for creating xmpp account for communication.

# The following two props assume we are using jigasi on the same machine as
# the xmpp server.
org.jitsi.jigasi.xmpp.acc.IS_SERVER_OVERRIDDEN=true
org.jitsi.jigasi.xmpp.acc.SERVER_ADDRESS=127.0.0.1
org.jitsi.jigasi.xmpp.acc.VIDEO_CALLING_DISABLED=true
org.jitsi.jigasi.xmpp.acc.JINGLE_NODES_ENABLED=false
org.jitsi.jigasi.xmpp.acc.AUTO_DISCOVER_STUN=false
org.jitsi.jigasi.xmpp.acc.IM_DISABLED=true
org.jitsi.jigasi.xmpp.acc.SERVER_STORED_INFO_DISABLED=true
org.jitsi.jigasi.xmpp.acc.IS_FILE_TRANSFER_DISABLED=true
org.jitsi.jigasi.xmpp.acc.KEEP_ALIVE_METHOD=XEP-0199
org.jitsi.jigasi.xmpp.acc.KEEP_ALIVE_INTERVAL=30
org.jitsi.jigasi.xmpp.acc.USE_DEFAULT_STUN_SERVER=false
# Or you can use bosh for the connection establishment by specifing the URL to use.
# org.jitsi.jigasi.xmpp.acc.BOSH_URL_PATTERN=https://server.com/http-bind?room={roomName}

#Used when outgoing calls are used in multidomain environment, used to detect subdomains
#org.jitsi.jigasi.xmpp.acc.DOMAIN_BASE=<<DOMAIN_BASE>>
#org.jitsi.jigasi.xmpp.acc.BOSH_URL_PATTERN=https://{host}{subdomain}/http-bind?room={roomName}

# can be enabled to disable audio mixing and use translator, jigasi will act as jvb, just forward every ssrc stream it receives.
#org.jitsi.jigasi.xmpp.acc.USE_TRANSLATOR_IN_CONFERENCE=true

# If you want jigasi to perform authenticated login instead of anonymous login
# to the XMPP server, you can set the following properties.
# org.jitsi.jigasi.xmpp.acc.USER_ID=SOME_USER@SOME_DOMAIN
# org.jitsi.jigasi.xmpp.acc.PASS=SOME_PASS
# org.jitsi.jigasi.xmpp.acc.ANONYMOUS_AUTH=false

# To fix SSL/TLS required by client but not supported by server
#org.jitsi.jigasi.xmpp.acc.ALLOW_NON_SECURE=true

# Can be used in combination with jitsi-meet module mod_auth_jitsi-shared-secret
# To have jigasi use a random username on every call
#org.jitsi.jigasi.xmpp.acc.UNIQUE_USER_ID=true

# If you want to disconnect jigasi calls automatically when all web users have
# left, you can set the following property to false.
# org.jitsi.jigasi.ALLOW_ONLY_JIGASIS_IN_ROOM=true

# If you want to use the SIP user part of the incoming/outgoing call SIP URI
# you can set the following property to true.
# org.jitsi.jigasi.USE_SIP_USER_AS_XMPP_RESOURCE=true

# Activate this property if you are using self-signed certificates or other
# type of non-trusted certicates. In this mode your service trust in the
# remote certificates always.
 net.java.sip.communicator.service.gui.ALWAYS_TRUST_MODE_ENABLED=true

# Enable this property to be able to shutdown gracefully jigasi using
# a rest command
# org.jitsi.jigasi.ENABLE_REST_SHUTDOWN=true

# Options regarding Transcription. Read the README for a detailed description
# about each property

#org.jitsi.jigasi.ENABLE_TRANSCRIPTION=false
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

# delivering final transcript
# org.jitsi.jigasi.transcription.DIRECTORY=/var/lib/jigasi/transcripts
# org.jitsi.jigasi.transcription.BASE_URL=http://localhost/
# 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

# send formats
# org.jitsi.jigasi.transcription.SEND_JSON=true
# org.jitsi.jigasi.transcription.SEND_TXT=false

# Vosk server
# org.jitsi.jigasi.transcription.customService=org.jitsi.jigasi.transcription.VoskTranscriptionService
# org.jitsi.jigasi.transcription.vosk.websocket_url={"en": "ws://localhost:2700", "fr": "ws://localhost:2710"}
# org.jitsi.jigasi.transcription.vosk.websocket_url=ws://localhost:2700

# Whisper live transcription server
# org.jitsi.jigasi.transcription.customService=org.jitsi.jigasi.transcription.WhisperTranscriptionService
# org.jitsi.jigasi.transcription.whisper.jwt_audience=jitsi
# org.jitsi.jigasi.transcription.whisper.private_key=<SINGLE LINE SSH-KEYGEN PRIVATE KEY>
# org.jitsi.jigasi.transcription.whisper.private_key_name=<A_UNIQUE_ID_FOR_THE_KEY_NO_SPACES>
# org.jitsi.jigasi.transcription.whisper.websocket_url=ws://localhost:8000/ws/

# Tenant-aware transcription service URL
# Allows defining which transcription service to use by doing a remote call.
# Expects '{"transcriber": "some-valid-transcriber"}'
# https://<domain>/tenant-added-when-starting-transcription
# org.jitsi.jigasi.transcription.remoteTranscriptionConfigUrl=<https://domain>

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

# translation
# org.jitsi.jigasi.transcription.ENABLE_TRANSLATION=false

# record audio. Currently only wav format is supported
# org.jitsi.jigasi.transcription.RECORD_AUDIO=false
# org.jitsi.jigasi.transcription.RECORD_AUDIO_FORMAT=wav

# execute one or more scripts when a transcript or recording is saved
# org.jitsi.jigasi.transcription.EXECUTE_SCRIPTS=true
# org.jitsi.jigasi.transcription.SCRIPTS_TO_EXECUTE_LIST_SEPARATOR=","
# org.jitsi.jigasi.transcription.SCRIPTS_TO_EXECUTE_LIST=script/example_handle_transcript_directory.sh

# filter out silent audio
#org.jitsi.jigasi.transcription.FILTER_SILENCE = false

# properties for optionally sending statistics to a DataDog server
#org.jitsi.ddclient.prefix=jitsi.jigasi
#org.jitsi.ddclient.host=localhost
#org.jitsi.ddclient.port=8125

# sip health checking
# Enables sip health checking by specifying a number/uri to call
# the target just needs to auto-connect the call play some audio,
# the call must be established for less than 10 seconds
# org.jitsi.jigasi.HEALTH_CHECK_SIP_URI=healthcheck
#
# The interval between healthcheck calls, by default is 5 minutes
# org.jitsi.jigasi.HEALTH_CHECK_INTERVAL=300000
#
# The timeout of healthcheck, if there was no successful health check for
# 10 minutes (default value) we consider jigasi unhealthy
# org.jitsi.jigasi.HEALTH_CHECK_TIMEOUT=600000

# Enabled or disable the notification when max occupants limit is reached
# org.jitsi.jigasi.NOTIFY_MAX_OCCUPANTS=false

[damencho]

Disable amr, do you see any changr? If not, what about when you disable g722?

[liangguohao66]

2025-04-14 02:35:08.991 INFO: [159] Aes.createCipher#433: Will employ AES implemented by SunJCE for AES/CTR/NoPadding.
2025-04-14 02:35:09.045 SEVERE: [220] net.sf.fmj.media.Log.error: Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
2025-04-14 02:35:09.046 SEVERE: [220] net.sf.fmj.media.Log.error: Failed to prefetch: net.sf.fmj.media.ProcessEngine@4f751e3e
2025-04-14 02:35:09.047 SEVERE: [219] net.sf.fmj.media.Log.error: Error: Unable to prefetch net.sf.fmj.media.ProcessEngine@4f751e3e

[liangguohao66]

Disable amr, do you see any changr? If not, what about when you disable g722?

not ok

[damencho]

How do you disable the codecs? Try them one by one, leaving everything with priority 0 and just one with some positive number. Is there a codec without the issue. You need to restart jigasi after config change.

[liangguohao66]

How do you disable the codecs? Try them one by one, leaving everything with priority 0 and just one with some positive number. Is there a codec without the issue. You need to restart jigasi after config change.

I commet with '#', not wok? will try priority 0

[liangguohao66]

How do you disable the codecs? Try them one by one, leaving everything with priority 0 and just one with some positive number. Is there a codec without the issue. You need to restart jigasi after config change.

opus is ok. thanks.

BTW, does the following error matter?

2025-04-15 03:34:33.420 SEVERE: [195] net.sf.fmj.media.Log.error: Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
2025-04-15 03:34:33.421 SEVERE: [195] net.sf.fmj.media.Log.error: Failed to prefetch: net.sf.fmj.media.ProcessEngine@58a29981
2025-04-15 03:34:33.422 SEVERE: [194] net.sf.fmj.media.Log.error: Error: Unable to prefetch net.sf.fmj.media.ProcessEngine@58a29981

[damencho]

Nope, you can ignore it.