jitsi docker stable-10078-1 PCMA error

[uholeschak]

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Description

In jitsi docker stable-10078-1 there is no sound transmitted in the outgoing direction. In the logs there is the error message:

Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed

Full log:

2025-03-04 15:11:00 2025-03-04 14:11:00.357 INFO: [146] SipGatewaySession$CallPeerListener.peerStateChanged#1577: [ctx=1741097456460611153864] SIP peer state: Connecting*
2025-03-04 15:11:00 2025-03-04 14:11:00.406 INFO: [170] net.java.sip.communicator.impl.protocol.sip.SipLogger.logInfo: Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /212.227.124.130:5060
2025-03-04 15:11:00 2025-03-04 14:11:00.407 INFO: [170] SipGatewaySession$SipCallStateListener.handleCallState#1504: [ctx=1741097456460611153864] Sip call IN_PROGRESS: Call: id=17410974564352141923083 peers=1
2025-03-04 15:11:00 2025-03-04 14:11:00.407 INFO: [170] SipGatewaySession$SipCallStateListener.handleCallState#1506: [ctx=1741097456460611153864] SIP call format used: rtpmap:8 PCMA/8000
2025-03-04 15:11:00 2025-03-04 14:11:00.408 INFO: [170] SipGatewaySession$CallPeerListener.peerStateChanged#1577: [ctx=1741097456460611153864] SIP peer state: Connected
2025-03-04 15:11:00 2025-03-04 14:11:00.410 INFO: [170] net.java.sip.communicator.service.protocol.media.CallPeerMediaHandler.start: Starting
2025-03-04 15:11:00 2025-03-04 14:11:00.518 INFO: [130] JvbConference$JvbCallChangeListener.callStateChangedInternal#1794: [ctx=1741097456460611153864] JVB conference call IN_PROGRESS.
2025-03-04 15:11:00 2025-03-04 14:11:00.538 SEVERE: [215] net.sf.fmj.media.Log.error:   Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
2025-03-04 15:11:00 2025-03-04 14:11:00.538 SEVERE: [215] net.sf.fmj.media.Log.error: Failed to prefetch: net.sf.fmj.media.ProcessEngine@7864a9f2
2025-03-04 15:11:00 2025-03-04 14:11:00.539 SEVERE: [211] net.sf.fmj.media.Log.error: Error: Unable to prefetch net.sf.fmj.media.ProcessEngine@7864a9f2
2025-03-04 15:11:00 
2025-03-04 15:11:00 2025-03-04 14:11:00.544 INFO: [170] net.java.sip.communicator.service.protocol.media.TransportManager.sendHolePunchPacket: Send NAT hole punch packets
2025-03-04 15:11:00 2025-03-04 14:11:00.571 INFO: [153] JitsiOpenSslProvider.<clinit>#52: jitsisrtp successfully loaded for OpenSSL 3
2025-03-04 15:11:00 2025-03-04 14:11:00.996 INFO: [153] Aes.benchmark#367: AES benchmark (of execution times expressed in nanoseconds): OpenSSL 769, SunJCE 3971, BouncyCastle 10711 for AES/CTR/NoPadding

The outgoing RTP data is send, but contains only constant values (silence). Checked with Wireshark.

Current behavior

Outgoing RTP data contains only constant data.

Expected Behavior

Outgoing RTP data contains valid data.

Possible Solution

No solution found.

Steps to reproduce

Create a SIP connection with PCMA LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed

Environment details

jitsi docker stable-10078-1 in Windows11.

sip-communicator.properties:

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

# 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

# Control REST Shutdown
org.jitsi.jigasi.ENABLE_REST_SHUTDOWN=false

org.jitsi.jigasi.BREWERY_ENABLED=true
org.jitsi.jigasi.MUC_JOIN_TIMEOUT=10

org.jitsi.jigasi.HEALTH_CHECK_SIP_URI=
org.jitsi.jigasi.HEALTH_CHECK_INTERVAL=30000
org.jitsi.jigasi.HEALTH_CHECK_TIMEOUT=600000

org.jitsi.jigasi.xmpp.acc.IS_SERVER_OVERRIDDEN=true
org.jitsi.jigasi.xmpp.acc.SERVER_ADDRESS=xmpp.meet.jitsi
org.jitsi.jigasi.xmpp.acc.SERVER_PORT=5222
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

# 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



org.jitsi.jigasi.DEFAULT_JVB_ROOM_NAME=hauskreis


org.jitsi.jigasi.MUC_SERVICE_ADDRESS=muc.meet.jitsi

# when checking other participants whether they are jibri/jigasi we can also check the the domain they use for connecting


org.jitsi.jigasi.JVB_INVITE_TIMEOUT=30000





# XMPP account used for control


net.java.sip.communicator.impl.protocol.jabber.acc-0=acc-0
net.java.sip.communicator.impl.protocol.jabber.acc-0.ACCOUNT_UID=Jabber:jigasi@auth.meet.jitsi@0
net.java.sip.communicator.plugin.reconnectplugin.ATLEAST_ONE_SUCCESSFUL_CONNECTION.Jabber\:jigasi@auth.meet.jitsi=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.USER_ID=jigasi@auth.meet.jitsi
net.java.sip.communicator.impl.protocol.jabber.acc-0.IS_SERVER_OVERRIDDEN=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.SERVER_ADDRESS=xmpp.meet.jitsi
net.java.sip.communicator.impl.protocol.jabber.acc-0.SERVER_PORT=5222
net.java.sip.communicator.impl.protocol.jabber.acc-0.ALLOW_NON_SECURE=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.PASSWORD=ZDYzNTk2YTA2NDEwYjIwZjY3MGM5ODA1MTQ0OWYxNmQ=
net.java.sip.communicator.impl.protocol.jabber.acc-0.AUTO_GENERATE_RESOURCE=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.RESOURCE_PRIORITY=30
net.java.sip.communicator.impl.protocol.jabber.acc-0.KEEP_ALIVE_METHOD=XEP-0199
net.java.sip.communicator.impl.protocol.jabber.acc-0.KEEP_ALIVE_INTERVAL=20
net.java.sip.communicator.impl.protocol.jabber.acc-0.CALLING_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.JINGLE_NODES_ENABLED=false
net.java.sip.communicator.impl.protocol.jabber.acc-0.IS_CARBON_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.DEFAULT_ENCRYPTION=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.IS_USE_ICE=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.IS_ACCOUNT_DISABLED=false
net.java.sip.communicator.impl.protocol.jabber.acc-0.IS_PREFERRED_PROTOCOL=false
net.java.sip.communicator.impl.protocol.jabber.acc-0.AUTO_DISCOVER_JINGLE_NODES=false
net.java.sip.communicator.impl.protocol.jabber.acc-0.PROTOCOL=Jabber
net.java.sip.communicator.impl.protocol.jabber.acc-0.IS_USE_UPNP=false
net.java.sip.communicator.impl.protocol.jabber.acc-0.IM_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.SERVER_STORED_INFO_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.IS_FILE_TRANSFER_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.USE_DEFAULT_STUN_SERVER=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.ENCRYPTION_PROTOCOL.DTLS-SRTP=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.ENCRYPTION_PROTOCOL_STATUS.DTLS-SRTP=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.VIDEO_CALLING_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.OVERRIDE_ENCODINGS=true
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.G722/8000=705
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.GSM/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.H263-1998/90000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.H264/90000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.PCMA/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.PCMU/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.SILK/12000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.SILK/16000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.SILK/24000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.SILK/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.VP8/90000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.iLBC/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.opus/48000=750
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.speex/16000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.speex/32000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.speex/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.telephone-event/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-0.BREWERY=jigasibrewery@internal-muc.meet.jitsi
net.java.sip.communicator.impl.protocol.jabber.acc-0.DOMAIN_BASE=meet.holeschak.de
org.jitsi.jigasi.ALLOWED_JID=jigasibrewery@internal-muc.meet.jitsi
org.jitsi.jigasi.ENABLE_SIP=true

# SIP account
net.java.sip.communicator.impl.protocol.sip.acc1=acc1

net.java.sip.communicator.impl.protocol.sip.acc1.PROXY_ADDRESS=sip.1und1.de
net.java.sip.communicator.impl.protocol.sip.acc1.PROXY_AUTO_CONFIG=false
net.java.sip.communicator.impl.protocol.sip.acc1.PROXY_PORT=5060
net.java.sip.communicator.impl.protocol.sip.acc1.PREFERRED_TRANSPORT=UDP


net.java.sip.communicator.impl.protocol.sip.acc1.ACCOUNT_UID=SIP\:4962132697243@sip.1und1.de
net.java.sip.communicator.impl.protocol.sip.acc1.PASSWORD=Sm9oYW5uZXMxMSEkPw==
net.java.sip.communicator.impl.protocol.sip.acc1.PROTOCOL_NAME=SIP
net.java.sip.communicator.impl.protocol.sip.acc1.SERVER_ADDRESS=sip.1und1.de
net.java.sip.communicator.impl.protocol.sip.acc1.USER_ID=4962132697243@sip.1und1.de
net.java.sip.communicator.impl.protocol.sip.acc1.KEEP_ALIVE_INTERVAL=25
net.java.sip.communicator.impl.protocol.sip.acc1.KEEP_ALIVE_METHOD=OPTIONS
net.java.sip.communicator.impl.protocol.sip.acc1.VOICEMAIL_ENABLED=false
net.java.sip.communicator.impl.protocol.sip.acc1.JITSI_MEET_ROOM_HEADER_NAME=X-Room-Name
net.java.sip.communicator.impl.protocol.sip.acc1.JITSI_MEET_DOMAIN_BASE_HEADER_NAME=X-Domain-Base
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.AMR-WB/16000=750
# Temporarily disable G722, see: https://github.com/jitsi/docker-jitsi-meet/issues/1357
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.G722/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.GSM/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.H263-1998/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.H264/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.PCMA/8000=600
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.PCMU/8000=650
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.SILK/12000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.SILK/16000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.SILK/24000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.SILK/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.VP8/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.iLBC/8000=10
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.opus/48000=1000
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.red/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.speex/16000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.speex/32000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.speex/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.telephone-event/8000=1
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.ulpfec/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1.OVERRIDE_ENCODINGS=true
net.java.sip.communicator.impl.protocol.sip.acc1.DOMAIN_BASE=meet.jitsi



org.jitsi.jigasi.xmpp.acc.USER_ID=jigasi@guest.meet.jitsi
org.jitsi.jigasi.xmpp.acc.ANONYMOUS_AUTH=true

org.jitsi.jigasi.xmpp.acc.PASS=d63596a06410b20f670c98051449f16d
org.jitsi.jigasi.xmpp.acc.ALLOW_NON_SECURE=true


# forces sending presence on sip call hangup
net.java.sip.communicator.impl.protocol.jabber.FORCE_PRESENCE_ON_LEAVE=true

[damencho]

Try changing the codec, does it work with something else? PCMU for example?

[uholeschak]

In custom-sip-communicator.properties I have tried to disable PCMA:

net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.PCMA/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.PCMU/8000=0

Then ist uses iLBC, what is extremely delayed, but still the same error message, and no sound in the outgoing direction.
Or is the error message related to the XMPP connection?
Is this related to #378 ?
Docker also uses Java 17

[damencho]

Have you tested a 3-way call (withot jigasi) on your deployment?

[uholeschak]

Yes video conferences with multiple users are working correctly.
The SIP connection is also created correctly and packets are send in both direction.
The only problem is there are constant values in the outgoing stream (silence).
Is the message the source of the problem?

Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed

[damencho]

I guess so.

[damencho]

Yeah, don't use ilbc. But ulaw should be working fine.

[uholeschak]

You mean setting only?

net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.PCMA/8000=0

I also tested this, same result ...
The strange thing is, the error message is always the same or does it come form the internal XMPP connection?

[damencho]

Nop, this is disabling it. Set every one else to 0 and just leave That one enabled.
Not sure what happens when you have a double setting in the props file is it the first or the last value used.

[uholeschak]

In the docker settings the default values are:

net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.PCMA/8000=600
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.PCMU/8000=650

So I if overwrite it with:

net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.PCMA/8000=0

PCMU is left over, which is ulaw I expect ...
Or what should I write in custom-sip-communicator.properties ?
The values of custom-sip-communicator.properties are simply copied at the end of sip-communicator.properties

[damencho]

So I if overwrite it with:

net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.PCMA/8000=0

When you do this, is it trying to use pcmu?

[uholeschak]

Yes but the error message is the same ...
This is the strange thing. Also with iLBC I get this message, which I did not expect.
The error message is not related to the selected codec.
A the moment I have simply no idea what causes the error message.
I also tried to change:

net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.PCMA/8000=600
net.java.sip.communicator.impl.protocol.jabber.acc-0.Encodings.PCMU/8000=650

Same result.

[damencho]

What codecs does the remote side support?
Yeah, I'm not sure about the problem. We do not use jigasi in this mode and we are not testing the transcoding case. I remember though there was a problem with alaw at some point ... ilbc is not something you should use at all. If that is enabled by default, we should turn it off.
We use jigasi in translator mode, basically the opus packets it receives from the bridge are being routed to the sip side, but in this mode you need to make sure your sip backend can handle multiple ssrc streams for the same call and is mixing them if needed.
We test though opus decoding with the transcriber mode, and that for sure works. So if your sip backend supports opus, just switch off everything and leave opus.

[uholeschak]

Opus is not supported, but you could see there is still the same error later.
The cause of the error must be something else.

2025-03-04 17:52:12 2025-03-04 16:52:12.915 INFO: [83] JvbConference$JvbCallListener.incomingCallReceivedInternal#1685: [ctx=1741107129290481575445] Got invite from focus
2025-03-04 17:52:12 2025-03-04 16:52:12.921 SEVERE: [100] net.java.sip.communicator.impl.protocol.sip.CallPeerSipImpl.answer: Failed to create an SDP description for an OK response to an INVITE request!
2025-03-04 17:52:12 net.java.sip.communicator.service.protocol.OperationFailedException: Offer contained no valid media descriptions.
2025-03-04 17:52:12 at net.java.sip.communicator.impl.protocol.sip.CallPeerMediaHandlerSipImpl.createMediaDescriptionsForAnswer(CallPeerMediaHandlerSipImpl.java:845)
2025-03-04 17:52:12 at net.java.sip.communicator.impl.protocol.sip.CallPeerMediaHandlerSipImpl.processFirstOffer(CallPeerMediaHandlerSipImpl.java:469)
2025-03-04 17:52:12 at net.java.sip.communicator.impl.protocol.sip.CallPeerMediaHandlerSipImpl.processOffer(CallPeerMediaHandlerSipImpl.java:432)
2025-03-04 17:52:12 at net.java.sip.communicator.impl.protocol.sip.CallPeerSipImpl.answer(CallPeerSipImpl.java:1339)
2025-03-04 17:52:12 at net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImpl.answerCallPeer(OperationSetBasicTelephonySipImpl.java:1953)
2025-03-04 17:52:12 at org.jitsi.jigasi.CallManager$AnswerCallThread.run(CallManager.java:356)
2025-03-04 17:52:12 at java.base/java.util.concurrent.Executors$RunnableAdapter.call(Executors.java:539)
2025-03-04 17:52:12 at java.base/java.util.concurrent.FutureTask.run(FutureTask.java:264)
2025-03-04 17:52:12 at java.base/java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1136)
2025-03-04 17:52:12 at java.base/java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:635)
2025-03-04 17:52:12 at java.base/java.lang.Thread.run(Thread.java:840)
2025-03-04 17:52:12 2025-03-04 16:52:12.928 INFO: [100] SipGatewaySession$SipCallStateListener.handleCallState#1516: [ctx=1741107129290481575445] SIP call ended: CallPeerChangeEvent: type=CallPeerStatusChange oldV=net.java.sip.communicator.service.protocol.CallPeerState:Incoming Call newV=net.java.sip.communicator.service.protocol.CallPeerState:Failed for peer=4962112286112 <4962112286112@sip.1und1.de>;status=Failed reason=Offer contained no valid media descriptions. reasonCode=-1
2025-03-04 17:52:12 2025-03-04 16:52:12.929 INFO: [100] SipGatewaySession$CallPeerListener.peerStateChanged#1577: [ctx=1741107129290481575445] SIP peer state: Failed
2025-03-04 17:52:12 2025-03-04 16:52:12.934 INFO: [104] net.java.sip.communicator.impl.protocol.sip.SipLogger.logInfo: Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /212.227.124.129:5060
2025-03-04 17:52:13 2025-03-04 16:52:13.023 INFO: [101] net.java.sip.communicator.service.protocol.media.MediaHandler.registerDynamicPTsWithStream: Dynamic PT map: 126=rtpmap:-1 telephone-event/8000; 111=rtpmap:-1 opus/48000/2 fmtp:useinbandfec=1;minptime=10; 
2025-03-04 17:52:13 2025-03-04 16:52:13.023 INFO: [101] net.java.sip.communicator.service.protocol.media.MediaHandler.registerDynamicPTsWithStream: PT overrides []
2025-03-04 17:52:13 2025-03-04 16:52:13.042 INFO: [101] net.java.sip.communicator.service.protocol.media.CallPeerMediaHandler.start: Starting
2025-03-04 17:52:13 2025-03-04 16:52:13.216 INFO: [83] JvbConference$JvbCallChangeListener.callStateChangedInternal#1794: [ctx=1741107129290481575445] JVB conference call IN_PROGRESS.
2025-03-04 17:52:14 2025-03-04 16:52:14.213 INFO: [112] JitsiOpenSslProvider.<clinit>#52: jitsisrtp successfully loaded for OpenSSL 3
2025-03-04 17:52:14 2025-03-04 16:52:14.568 INFO: [112] Aes.benchmark#367: AES benchmark (of execution times expressed in nanoseconds): OpenSSL 655, SunJCE 563, BouncyCastle 9571 for AES/CTR/NoPadding
2025-03-04 17:52:14 2025-03-04 16:52:14.568 INFO: [112] Aes.createCipher#433: Will employ AES implemented by SunJCE for AES/CTR/NoPadding.
2025-03-04 17:52:14 2025-03-04 16:52:14.608 SEVERE: [152] net.sf.fmj.media.Log.error:   Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
2025-03-04 17:52:14 2025-03-04 16:52:14.608 SEVERE: [152] net.sf.fmj.media.Log.error: Failed to prefetch: net.sf.fmj.media.ProcessEngine@5804d448
2025-03-04 17:52:14 2025-03-04 16:52:14.609 SEVERE: [151] net.sf.fmj.media.Log.error: Error: Unable to prefetch net.sf.fmj.media.ProcessEngine@5804d448

[damencho]

2025-03-04 17:52:12 2025-03-04 16:52:12.921 SEVERE: [100] net.java.sip.communicator.impl.protocol.sip.CallPeerSipImpl.answer: Failed to create an SDP description for an OK response to an INVITE request!
2025-03-04 17:52:12 net.java.sip.communicator.service.protocol.OperationFailedException: Offer contained no valid media descriptions.

Yep, this is because opus is not supported.

[uholeschak]

But why is there still the message?

2025-03-04 17:52:14 2025-03-04 16:52:14.608 SEVERE: [152] net.sf.fmj.media.Log.error:   Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed

This seems a general error ...