SIP users are hidden

I’ve installed jitsi-meet tested it, all worked as expected. Then I followed directions to enable LDAP authentication. LDAP auth works fine and I’m able to do videoconferences between users. I’ve also set up jigasi and now can dial into a conference room (from Freeswitch, albeit using a downgraded version), and can also dial out to add a user. So far so good…except no avatars, icons or announcements appear to indicate the SIP users are in the conference (but they did before I enabled LDAP)…as if they’re hidden. What did I miss (or mess up)? How do I enable the SIP users to be seen?

Do you have hiddenDomain setting in your config.js?

Nope, only place that hiddenDomain appears is in the commented out list of undocumented settings. SIP users were working “normally” (i.e. connections announced, avatars appeared, participant count incremented, etc) before applying the LDAP settings found here: (which I lifted from a previous forum discussion).

Any other thoughts/clues on this? I’ve rebuilt several times now and it’s 100% reproducible…

I’ve checked all of the various log files, and the only one that seems mildly unhappy is JVB. I don’t know enough about this to know if this is even related, and since the SIP call is actually up and functional (i.e. two way audio works to the bridge), I’m not sure how relevant this is but this is the only error/warning across all logs:

JVB 2020-02-18 03:32:09.398 WARNING: [2807] org.jitsi.videobridge.EndpointMessageTransport.log() SCTP connection with stewart not ready yet.
JVB 2020-02-18 03:32:09.398 WARNING: [2807] org.jitsi.videobridge.EndpointMessageTransport.log() No available transport channel, can’t send a message

I’m not sure how this is possible. So Jigasi enters a xmpp muc, if it doesn’t succeed you will see it in the jigasi logs and no audio will be possible. If it enters the muc, jicofo sends the invite so the media session can be established. Other participants see the jigasi participant and show the thumbnail. The only case when the thumbnail is not shown inn the UI of the clients is when jigasi uses the domain to authenticate which is set as hidden domain, normally this is used for jibri so we can hide the jibri participant.

I’m not sure if my understanding of this is correct, but I believed my jigasi is authenticating using the org.jitsi.jigasi.xmpp.acc.USER_ID, which is currently set to (where host.mydomain.tld is actually my real FQDN for the host). Pretty much, other than adding an anonymous domain ( and changing the authentication to ldap2 for my host.mydomain.tld, all of the domain stuff has been left at defaults. Other than your previous suggestion that hiddenDomain was set somewhere (reasonably sure it’s not), is there some other way for a domain to become hidden? Where should I look. Or am I somehow authenticating with the wrong credentials?

Also, I do note a line in my jigasi log that says Authenticated: false, but it seems to join the room anyways…(while on the web side of things, I’m prompted for ldap credentials). Here is a redacted (for domain name) portion of my jigasi.log…does this give you any further clues?

2020-02-20 16:09:57.997 INFO: [11650] org.jitsi.jigasi.SipGateway.incomingCallReceived().188 Incoming call received...
2020-02-20 16:09:58.998 INFO: [11651] Using default JVB room name property siptest
2020-02-20 16:09:59.000 INFO: [11651] org.jitsi.jigasi.JvbConference.setXmppProvider().567 7a99eae8@host.mydomain.tld will use ProtocolProviderServiceJabberImpl( (Jabber))
2020-02-20 16:09:59.060 INFO: [11653] impl.protocol.jabber.OperationSetBasicTelephonyJabberImpl.registrationStateChanged().125 Jingle : ON
2020-02-20 16:09:59.061 INFO: [11653] org.jitsi.jigasi.JvbConference.registrationStateChanged().612 XMPP (7a99eae8@host.mydomain.tld): RegistrationStateChangeEvent[ oldState=Registering; newState=RegistrationState=Registering; reasonCode=-1; reason=null]
2020-02-20 16:09:59.075 INFO: [11653] impl.protocol.jabber.ProtocolProviderServiceJabberImpl.authenticated().2535 Authenticated: false
2020-02-20 16:09:59.076 INFO: [11653] org.jitsi.jigasi.JvbConference.joinConferenceRoom().653 Joining JVB conference room:
2020-02-20 16:09:59.082 INFO: [11656] impl.protocol.jabber.ChatRoomJabberImpl.joined().1256 has joined the chat room.
2020-02-20 16:09:59.093 INFO: [11656] impl.protocol.jabber.ChatRoomJabberImpl.joined().1256 has joined the chat room.
2020-02-20 16:09:59.150 INFO: [11662] impl.protocol.jabber.IceUdpTransportManager.createIceAgent().254 Auto discovered harvester is null
2020-02-20 16:09:59.151 INFO: [11662] impl.protocol.jabber.IceUdpTransportManager.createIceAgent().346 End gathering harvester within 1 ms
2020-02-20 16:10:00.262 INFO: [11662] impl.protocol.jabber.CallPeerMediaHandlerJabberImpl.harvestCandidates().1198 End candidate harvest within 1111 ms
2020-02-20 16:10:00.266 INFO: [11662] org.jitsi.jigasi.JvbConference.incomingCallReceived().990 Got invite from focus
2020-02-20 16:10:00.309 INFO: [92] 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;
2020-02-20 16:10:00.310 INFO: [92] PT overrides [103->104 ]
2020-02-20 16:10:00.311 INFO: [92] Starting
2020-02-20 16:10:00.356 INFO: [92] org.jitsi.jigasi.JvbConference.onJvbCallStarted().771 JVB conference call IN_PROGRESS siptest
2020-02-20 16:10:00.357 INFO: [92] org.jitsi.jigasi.JvbConference.peerStateChanged().1056 7a99eae8@host.mydomain.tld JVB peer state:
2020-02-20 16:10:00.357 INFO: [92] org.jitsi.jigasi.JvbConference.advertisePeerSSRCs().298 Peer SSRCs audio: 432076847 video: null
2020-02-20 16:10:00.368 INFO: [275] Dynamic PT map: 101=rtpmap:-1 telephone-event/8000;
2020-02-20 16:10:00.368 INFO: [275] PT overrides []
2020-02-20 16:10:00.383 INFO: [275] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().1007 7a99eae8@host.mydomain.tld SIP peer state: Connecting*
2020-02-20 16:10:00.388 INFO: [11708] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /
2020-02-20 16:10:00.390 INFO: [11708] org.jitsi.jigasi.SipGatewaySession.handleCallState().929 Sip call IN_PROGRESS: Call: id=15822149979961608369635 peers=1
2020-02-20 16:10:00.390 INFO: [11708] org.jitsi.jigasi.SipGatewaySession.handleCallState().937 SIP call format used: rtpmap:0 PCMU/8000
2020-02-20 16:10:00.391 INFO: [11708] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().1007 7a99eae8@host.mydomain.tld SIP peer state: Connected
2020-02-20 16:10:00.393 INFO: [11708] Starting
2020-02-20 16:10:00.397 SEVERE: [11723]   Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
2020-02-20 16:10:00.397 SEVERE: [11723] Failed to prefetch:
2020-02-20 16:10:00.398 SEVERE: [11721] Error: Unable to prefetch

2020-02-20 16:10:00.416 INFO: [11708] Send NAT hole punch packets
2020-02-20 16:10:01.302 INFO: [11763] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /
2020-02-20 16:10:03.387 INFO: [11764] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /

That is your problem, I’m not sure how it works at all. Focus is username that is reserved and used only by jicofo, you should not reuse it.

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Thank you, that was it! Created a new user, used that instead and all works as expected now (at least as far as SIP users being announced). I really appreciate your help!