Jigasi, no audio when other are mute (even with NAT hole punch and nat=force_rport,comedia)

[drali]

hi,

i am using freeBPX and asterisk. asterisk and Jigasi are inside separate dockers.
when i call in Jigasi, it can connect to room but if every other participants are mute, they don’t have Jigasi’s audio. once someone unmute it’s mic, everyone can hear Jigasi.

I thought NAT hole punch will solve it, but it dose not.

from Jigasi log:
Jigasi 2021-04-06 03:23:50.848 INFO: [217] service.protocol.media.TransportManager.sendHolePunchPacket().552 Send NAT hole punch packets

and in astrisk i have nat=force_rport,comedia config for all extensions.

i am using chan_sip, i tried pjsip and it didn’t worked too.

thanks for your help

[damencho]

Can you try adding a jigasi config net.java.sip.communicator.impl.protocol.HOLE_PUNCH_PKT_COUNT= maybe try with 6 or even 24 … and see does that change it?
I think, sometimes hole-punch packets can be dropped especially if the machine is a little bit loaded …

[drali]

no, nothing changed
i think something is wrong in freepbx config i attached some photos too

my jigasi log
Jigasi 2021-04-06 19:22:21.684 INFO: [317] org.jitsi.jigasi.JvbConference.incomingCallReceived().1380 [ctx=16177207411152095593970] Got invite from focus
Jigasi 2021-04-06 19:22:21.688 INFO: [99] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1003 Dynamic PT map: 101=rtpmap:-1 telephone-event/8000;
Jigasi 2021-04-06 19:22:21.688 INFO: [99] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1020 PT overrides
Jigasi 2021-04-06 19:22:21.689 INFO: [99] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().1682 [ctx=16177207411152095593970] SIP peer state: Connecting*
Jigasi 2021-04-06 19:22:21.690 INFO: [325] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /[myserverip]:5160
Jigasi 2021-04-06 19:22:21.690 INFO: [325] org.jitsi.jigasi.SipGatewaySession.handleCallState().1599 [ctx=16177207411152095593970] Sip call IN_PROGRESS: Call: id=16177207411141980781186 peers=1
Jigasi 2021-04-06 19:22:21.690 INFO: [325] org.jitsi.jigasi.SipGatewaySession.handleCallState().1608 [ctx=16177207411152095593970] SIP call format used: rtpmap:8 PCMA/8000
Jigasi 2021-04-06 19:22:21.691 INFO: [325] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().1682 [ctx=16177207411152095593970] SIP peer state: Connected
Jigasi 2021-04-06 19:22:21.691 INFO: [325] service.protocol.media.CallPeerMediaHandler.start().1961 Starting
Jigasi 2021-04-06 19:22:21.704 INFO: [325] service.protocol.media.TransportManager.sendHolePunchPacket().552 Send NAT hole punch packets
Jigasi 2021-04-06 19:22:21.747 INFO: [322] 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;
Jigasi 2021-04-06 19:22:21.747 INFO: [322] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1020 PT overrides [103->104 ]
Jigasi 2021-04-06 19:22:21.749 INFO: [322] service.protocol.media.CallPeerMediaHandler.start().1961 Starting
Jigasi 2021-04-06 19:22:21.775 INFO: [322] org.jitsi.jigasi.JvbConference.callStateChanged().1491 [ctx=16177207411152095593970] JVB conference call IN_PROGRESS.
Jigasi 2021-04-06 19:22:23.690 SEVERE: [326] org.jitsi.jigasi.SipGatewaySession.run().1533 [ctx=16177207411152095593970] Stopped receiving RTP for Call: id=16177207411141980781186 peers=1

-----> where i unmte the mic

Jigasi 2021-04-06 19:22:34.437 INFO: [378] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /[myserverip]:5160
Jigasi 2021-04-06 19:22:36.365 SEVERE: [401] net.sf.fmj.media.Log.error() Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
Jigasi 2021-04-06 19:22:36.365 SEVERE: [401] net.sf.fmj.media.Log.error() Failed to prefetch: net.sf.fmj.media.ProcessEngine@5128f55e
Jigasi 2021-04-06 19:22:36.366 SEVERE: [399] net.sf.fmj.media.Log.error() Error: Unable to prefetch net.sf.fmj.media.ProcessEngine@5128f55e
Jigasi 2021-04-06 19:22:37.690 INFO: [326] org.jitsi.jigasi.SipGatewaySession.run().1559 [ctx=16177207411152095593970] RTP resumed for Call: id=16177207411141980781186 peers=1
Jigasi 2021-04-06 19:22:41.823 INFO: [424] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /[myserverip]:5160

and my freebpx config

[drali]

i changed this option and now seems it is ok

[damencho]

Interesting, good to know, thank you.
Do which asterisk config is this?

[damencho]

Saw it it is in rtp.conf strictrtp=no.

Hum, I wonder why asterisk drop it though …

[drali]

thanks for your help, as this problem solved can you help me here too?

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[Murray26]

The ones which had video empowered didn’t encounter the issue, those with sound just could some of the time not hear any remaining members!