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

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 :slight_smile:

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 …

1 Like

no, nothing changed
i think something is wrong in freepbx config :frowning: 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


i changed this option and now seems it is ok
Screenshot from 2021-04-06 20-41-33

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

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

Hum, I wonder why asterisk drop it though …

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

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!