SIP connection drops after conference with dial-in user finishes

Issue: after every meeting that has a dial-in participant using Jigasi number, the SIP connection terminates unexpectedly right at the end of the call, and never recovers. It seems, it is trying to pass traffic, but the remote server never receives it. It tries for a few minutes, then gives up permanently. Further, I have to restart all jitsi services (jvb, jicofo, jigasi, prosody) several times to get it to reconnect.

Relevant lines (jigasi.log):

2020-05-11 11:34:05.294 INFO: [36] org.jitsi.jigasi.SipGateway.registrationStateChanged().105 REG STATE CHANGE ProtocolProviderServiceSipImpl(SIP:9566@voip.mydomain.com) -> RegistrationStateChangeEvent[ oldState=Unregistered; newState=RegistrationState=Registering; userRequest=false; reasonCode=-1; reason=null]
2020-05-11 11:34:37.303 INFO: [36285] org.jitsi.jigasi.SipGateway.registrationStateChanged().105 REG STATE CHANGE ProtocolProviderServiceSipImpl(SIP:9566@voip.mydomain.com) -> RegistrationStateChangeEvent[ oldState=Registering; newState=RegistrationState=Connection Failed; userRequest=false; reasonCode=-1; reason=A timeout occurred while trying to connect to the server.]

Symptoms:

  • SIP traffic continues for about 2 more minutes, then gives up.
  • Remote server has no record of the attempts following the closed call.
  • This traffic is passing thru a VPN tunnel between 10.x.x.x and 192.168.129.4, established via Strongswan VPN; I doubt this is related but worth mentioning

Theory:

  • There may be a bug that causes the socket to become invalid when the call terminates, and it gets in a stuck state.

Already tried/didn’t work

  • net.java.sip.communicator.plugin.reconnectplugin.ATLEAST_ONE_SUCCESSFUL_CONNECTION.acc1403273890647=true
  • I’m sure the FreePBX voip server is configured and working, because it works after simply restarting all the Jitsi services.

You can help me by:

  • Telling me how I can instruct the serve to continue attempting to connect indefinitely, no matter how many failures there are.
  • Telling me how I can add more verbosity to my logs to give me details.
  • Suggesting properties to set in sip-communicator.properties that might fix the issue
  • Pointing me to places to look in other services logs (jvb, jicofo, prosody etc) and what to look for.

jigasi.log (11.7 KB) sip-communicator.properties.txt (8.7 KB)