Jigasi - No audio and call interrupts after random seconds

Hi everyone,

I have an Asterisk 13/FreePBX 15 connected to Sipgate basic. And have my jigasi connected to Asterisk/FreePBX.

The call on my Sipgate number is signaled on jigasi and will redirected to the room “siptest”. This works. I can see, that the phone is calling and connecting in the jitsi-meet website.

But after some random seconds the call ends. And I have no sound at the whole time in any direction to or from the phone.

Here is the jigasi logfile during the call:

2020-04-30 10:51:28.809 INFO: [189] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /192.168.16.89:5060
2020-04-30 10:51:31.686 INFO: [190] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /192.168.16.89:5060
2020-04-30 10:51:31.688 INFO: [190] org.jitsi.jigasi.SipGateway.incomingCallReceived().188 Incoming call received...
2020-04-30 10:51:32.689 INFO: [191] org.jitsi.jigasi.SipGatewaySession.run().907 Using default JVB room name property siptest
2020-04-30 10:51:32.691 INFO: [191] org.jitsi.jigasi.JvbConference.setXmppProvider().561 171ca48a8e8@test-jitsi.domain.tld will use ProtocolProviderServiceJabberImpl(171ca48a8e8@test-jitsi.domain.tld (Jabber))
2020-04-30 10:51:32.753 INFO: [193] impl.protocol.jabber.OperationSetBasicTelephonyJabberImpl.registrationStateChanged().125 Jingle : ON
2020-04-30 10:51:32.753 INFO: [193] org.jitsi.jigasi.JvbConference.registrationStateChanged().606 XMPP (171ca48a8e8@test-jitsi.domain.tld): RegistrationStateChangeEvent[ oldState=Registering; newState=RegistrationState=Registering; reasonCode=-1; reason=null]
2020-04-30 10:51:32.757 INFO: [193] impl.protocol.jabber.ProtocolProviderServiceJabberImpl.authenticated().2536 Authenticated: false
2020-04-30 10:51:32.759 INFO: [193] org.jitsi.jigasi.JvbConference.joinConferenceRoom().647 Joining JVB conference room: siptest
2020-04-30 10:51:32.780 INFO: [197] impl.protocol.jabber.ChatRoomJabberImpl.joined().1247 siptest@conference.test-jitsi.domain.tld/focus has joined the siptest@conference.test-jitsi.domain.tld chat room.
2020-04-30 10:51:32.781 INFO: [197] impl.protocol.jabber.ChatRoomJabberImpl.joined().1247 siptest@conference.test-jitsi.domain.tld/f1b534fa has joined the siptest@conference.test-jitsi.domain.tld chat room.
2020-04-30 10:51:32.783 INFO: [197] impl.protocol.jabber.ChatRoomJabberImpl.joined().1247 siptest@conference.test-jitsi.domain.tld/171ca48a8e8 has joined the siptest@conference.test-jitsi.domain.tld chat room.
2020-04-30 10:51:32.833 INFO: [202] impl.protocol.jabber.IceUdpTransportManager.createIceAgent().254 Auto discovered harvester is null
2020-04-30 10:51:32.860 INFO: [202] impl.protocol.jabber.IceUdpTransportManager.createIceAgent().346 End gathering harvester within 28 ms
2020-04-30 10:51:33.969 INFO: [202] impl.protocol.jabber.CallPeerMediaHandlerJabberImpl.harvestCandidates().1198 End candidate harvest within 1108 ms
2020-04-30 10:51:33.977 INFO: [202] org.jitsi.jigasi.JvbConference.incomingCallReceived().965 Got invite from focus
2020-04-30 10:51:34.034 INFO: [207] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1003 Dynamic PT map: 126=rtpmap:-1 telephone-event/8000; 111=rtpmap:-1 opus/48000/2 fmtp:useinbandfec=1;minptime=10; 103=rtpmap:-1 unknown/90000;
2020-04-30 10:51:34.034 INFO: [207] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1020 PT overrides [103->104 ]
2020-04-30 10:51:34.042 INFO: [207] service.protocol.media.CallPeerMediaHandler.start().1960 Starting
2020-04-30 10:51:34.084 INFO: [207] org.jitsi.jigasi.JvbConference.onJvbCallStarted().746 JVB conference call IN_PROGRESS siptest
2020-04-30 10:51:34.086 INFO: [207] org.jitsi.jigasi.JvbConference.peerStateChanged().1031 171ca48a8e8@test-jitsi.domain.tld JVB peer state: net.java.sip.communicator.service.protocol.CallPeerState:Connected
2020-04-30 10:51:34.086 INFO: [207] org.jitsi.jigasi.JvbConference.advertisePeerSSRCs().292 Peer net.java.sip.communicator.service.protocol.CallPeerState:Connected SSRCs audio: 840903211 video: null
2020-04-30 10:51:34.111 INFO: [228] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1003 Dynamic PT map: 101=rtpmap:-1 telephone-event/8000;
2020-04-30 10:51:34.111 INFO: [228] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1020 PT overrides []
2020-04-30 10:51:34.113 INFO: [228] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().828 171ca48a8e8@test-jitsi.domain.tld SIP peer state: Connecting*
2020-04-30 10:51:34.118 INFO: [239] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /192.168.16.89:5060
2020-04-30 10:51:34.119 INFO: [239] org.jitsi.jigasi.SipGatewaySession.handleCallState().753 Sip call IN_PROGRESS: Call: id=15882366916871729285813 peers=1
2020-04-30 10:51:34.119 INFO: [239] org.jitsi.jigasi.SipGatewaySession.handleCallState().761 SIP call format used: rtpmap:0 PCMU/8000
2020-04-30 10:51:34.120 INFO: [239] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().828 171ca48a8e8@test-jitsi.domain.tld SIP peer state: Connected
2020-04-30 10:51:34.121 INFO: [239] service.protocol.media.CallPeerMediaHandler.start().1960 Starting
2020-04-30 10:51:34.153 SEVERE: [273] net.sf.fmj.media.Log.error()   Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
2020-04-30 10:51:34.153 SEVERE: [273] net.sf.fmj.media.Log.error() Failed to prefetch: net.sf.fmj.media.ProcessEngine@3d98858f
2020-04-30 10:51:34.157 SEVERE: [272] net.sf.fmj.media.Log.error() Error: Unable to prefetch net.sf.fmj.media.ProcessEngine@3d98858f

2020-04-30 10:51:34.176 INFO: [239] service.protocol.media.TransportManager.sendHolePunchPacket().552 Send NAT hole punch packets
2020-04-30 10:51:41.251 INFO: [288] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /192.168.16.89:5060
2020-04-30 10:52:06.254 INFO: [291] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /192.168.16.89:5060
2020-04-30 10:52:28.809 INFO: [294] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /192.168.16.89:5060
2020-04-30 10:52:31.252 INFO: [295] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /192.168.16.89:5060
2020-04-30 10:52:39.463 INFO: [296] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /192.168.16.89:5060
2020-04-30 10:52:39.465 INFO: [296] org.jitsi.jigasi.SipGatewaySession.handleCallState().766 SIP call ended: CallPeerChangeEvent: type=CallPeerStatusChange oldV=net.java.sip.communicator.service.protocol.CallPeerState:Connected newV=net.java.sip.communicator.service.protocol.CallPeerState:Disconnected for peer= <01711234567@192.168.16.89>;status=Disconnected
2020-04-30 10:52:39.466 INFO: [296] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().828 171ca48a8e8@test-jitsi.domain.tld SIP peer state: Disconnected
2020-04-30 10:52:44.466 INFO: [297] org.jitsi.jigasi.SipGatewaySession.sipCallEnded().507 Sip call ended: Call: id=15882366916871729285813 peers=0
2020-04-30 10:52:44.468 INFO: [297] org.jitsi.jigasi.JvbConference.stop().521 171ca48a8e8@test-jitsi.domain.tld is removing account Jabber:171ca48a8e8@test-jitsi.domain.tld/171ca48a8e8
2020-04-30 10:52:44.475 SEVERE: [209] org.jitsi.impl.neomedia.RTPConnectorOutputStream.log() Failed to send a packet to target /194.95.1.2:10000:java.io.IOException: No active socket.
2020-04-30 10:52:44.480 INFO: [297] impl.protocol.jabber.OperationSetBasicTelephonyJabberImpl.registrationStateChanged().132 Jingle : OFF
2020-04-30 10:52:44.480 INFO: [96] org.jitsi.jigasi.JvbConference.peerStateChanged().1031 171ca48a8e8@test-jitsi.domain.tld JVB peer state: net.java.sip.communicator.service.protocol.CallPeerState:Disconnected
2020-04-30 10:52:44.480 INFO: [297] org.jitsi.jigasi.AbstractGateway.notifyCallEnded().141 Removed session for call 171ca48a8e8@test-jitsi.domain.tld
2020-04-30 10:52:44.494 SEVERE: [208] org.jitsi.impl.neomedia.RTPConnectorOutputStream.log() Failed to send a packet to target /194.95.1.2:10000:java.io.IOException: No active socket.
2020-04-30 10:52:44.496 SEVERE: [96] org.jitsi.impl.neomedia.RTPConnectorOutputStream.log() Failed to send a packet to target /194.95.1.2:10000:java.io.IOException: No active socket.

In the last days I learned, that the most interesting lines in the logfile are the SEVERE lines:

2020-04-30 10:51:34.153 SEVERE: [273] net.sf.fmj.media.Log.error()   Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
2020-04-30 10:51:34.153 SEVERE: [273] net.sf.fmj.media.Log.error() Failed to prefetch: net.sf.fmj.media.ProcessEngine@3d98858f
2020-04-30 10:51:34.157 SEVERE: [272] net.sf.fmj.media.Log.error() Error: Unable to prefetch net.sf.fmj.media.ProcessEngine@3d98858f
2020-04-30 10:52:44.475 SEVERE: [209] org.jitsi.impl.neomedia.RTPConnectorOutputStream.log() Failed to send a packet to target /194.95.1.2:10000:java.io.IOException: No active socket.
2020-04-30 10:52:44.494 SEVERE: [208] org.jitsi.impl.neomedia.RTPConnectorOutputStream.log() Failed to send a packet to target /194.95.1.2:10000:java.io.IOException: No active socket.
2020-04-30 10:52:44.496 SEVERE: [96] org.jitsi.impl.neomedia.RTPConnectorOutputStream.log() Failed to send a packet to target /194.95.1.2:10000:java.io.IOException: No active socket.

On the IP is on port 10000 the bridge listening.

I am using jigasi version 1.1-101-g3b2a0e5-1 but I also tried with version 1.0-235.

The system is a Debian 10 and the other jitsi versions are:
ii jitsi-meet-prosody 1.0.4025-1 all Prosody configuration for Jitsi Meet
ii jitsi-meet-web 1.0.4025-1 all WebRTC JavaScript video conferences
ii jitsi-meet-web-config 1.0.4025-1 all Configuration for web serving of Jitsi Meet
ii jitsi-videobridge2 2.1-183-gdbddd169-1 all WebRTC compatible Selective Forwarding Unit (SFU)

Has someone a hint for me?

Short update:

I changed the SIP connections from asterisk directly to my sipgate basic account. Then I have sound.
But this errors are the same:

2020-04-30 11:10:56.314 SEVERE: [250] impl.protocol.sip.SipStackSharing.processRequest().709 couldn’t find a ProtocolProviderServiceSipImpl to dispatch to
2020-04-30 11:10:57.077 SEVERE: [327] net.sf.fmj.media.Log.error() Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
2020-04-30 11:10:57.077 SEVERE: [327] net.sf.fmj.media.Log.error() Failed to prefetch: net.sf.fmj.media.ProcessEngine@57de3d32
2020-04-30 11:10:57.079 SEVERE: [326] net.sf.fmj.media.Log.error() Error: Unable to prefetch net.sf.fmj.media.ProcessEngine@57de3d32
2020-04-30 11:11:00.314 SEVERE: [351] impl.protocol.sip.SipStackSharing.findTargetFor().922 no listeners
2020-04-30 11:11:00.314 SEVERE: [351] impl.protocol.sip.SipStackSharing.processRequest().709 couldn’t find a ProtocolProviderServiceSipImpl to dispatch to