Jitsi-Conference-Room doesn't seem to be read

Hi,

I’m sending the following invite to Jigasi:

INVITE sip:9999990@a.b.c.d:5060 SIP/2.0
Via: SIP/2.0/UDP a.b.c.189:5070;branch=z9hG4bK7bb9ebb1;rport
Max-Forwards: 70
From: <sip:0878209509@a.b.c.189:5070>;tag=as62ca4d27
To: <sip:9999990@a.b.c.d:5060>
Contact: <sip:0878209509@a.b.c.189:5070>
Call-ID: 6570b57c39c288c7733219f050107a64@a.b.c.189:5070
CSeq: 102 INVITE
User-Agent: Telviva N+
Date: Fri, 15 May 2020 16:14:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Jitsi-Conference-Room: 91-StevesConf1472303
Diversion: <sip:1234554321@ct-dev01>
Content-Type: application/sdp
Content-Length: 374
...

Jigasi sends 180 Ringing and then a 486 Busy here

Jigasi log says:

Jigasi 2020-05-15 18:14:00.292 INFO: [50] org.jitsi.jigasi.SipGateway.incomingCallReceived().196 [ctx=1589559240291349186763] Incoming call received...
Jigasi 2020-05-15 18:14:01.292 INFO: [74] org.jitsi.jigasi.SipGatewaySession.run().1515 [ctx=1589559240291349186763] No JVB room name provided in INVITE header
Jigasi 2020-05-15 18:14:01.294 INFO: [75] org.jitsi.jigasi.SipGatewaySession.handleCallState().1391 [ctx=1589559240291349186763] SIP call ended: CallPeerChangeEvent: type=CallPeerStatusChange oldV=net.java.sip.communicator.service.protocol.CallPeerState:Incoming Call newV=net.java.sip.communicator.service.protocol.CallPeerState:Failed for peer=0878209509 <0878209509@a.b.c.189>;status=Failed
Jigasi 2020-05-15 18:14:01.294 INFO: [75] org.jitsi.jigasi.SipGatewaySession.sipCallEnded().584 [ctx=1589559240291349186763] Sip call ended: Call: id=15895592402911797930361 peers=0
Jigasi 2020-05-15 18:14:01.294 SEVERE: [75] org.jitsi.jigasi.AbstractGateway.notifyCallEnded().120 [ctx=1589559240291349186763] Call resource not exists for session.
Jigasi 2020-05-15 18:14:01.295 INFO: [75] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().1457 null SIP peer state: Failed

Why does it say No JVB room name provided in INVITE header when I do have the Jitsi-Conference-Room header as per the docs at https://github.com/jitsi/jigasi#incoming-calls ?

I have that room and was connected to it from Jitsi-meet at the time.

Help is appreciated!

Hi,

For completeness, here is the complete Dialog as captured on the outside of my Kamailio proxy and on the way to jigasi.

There is no further proxy or anything between here and jigasi, so the header must have arrived on Jigasi

T a.b.c.189:5060 -> 13.x.y.z:44557 [AP]
INVITE sip:9999990@172.17.0.2:44557;transport=tcp;registering_acc=vc2_xxxxx_com SIP/2.0.
Record-Route: <sip:a.b.c.189;transport=tcp;r2=on;lr=on>.
Record-Route: <sip:a.b.c.189;r2=on;lr=on>.
Via: SIP/2.0/TCP a.b.c.189;branch=z9hG4bKa4d7.55e5aded836f1ad498a6289d5083fa44.0.
Via: SIP/2.0/UDP a.b.c.189:5070;received=a.b.c.189;branch=z9hG4bK05a2b5f0;rport=5070.
Max-Forwards: 69.
From: <sip:0878209509@a.b.c.189:5070>;tag=as101814de.
To: <sip:9999990@a.b.c.189:5060>.
Contact: <sip:0878209509@a.b.c.189:5070>.
Call-ID: 5e67085670baa1be07d898c605240dfd@a.b.c.189:5070.
CSeq: 102 INVITE.
User-Agent: Telviva N+.
Date: Fri, 15 May 2020 16:37:40 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
X-Enswitch-Server: .
X-Enswitch-Uniqueid: .
X-Enswitch-Callid: 1589560654.47836285.
Jitsi-Conference-Room: 91-StevesConf1472303.
Diversion: <sip:1234554321@ct-dev01>.
Content-Type: application/sdp.
Content-Length: 372.
X-Enswitch-RURI: sip:9999990@a.b.c.189:5060.
X-Enswitch-Source: a.b.c.189:5070.
.
v=0.
o=root 175264678 175264678 IN IP4 a.b.c.189.
s=Telviva.
c=IN IP4 a.b.c.189.
t=0 0.
m=audio 10806 RTP/AVP 9 117 8 18 110 101.
a=rtpmap:9 G722/8000.
a=rtpmap:117 speex/16000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:110 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=maxptime:60.
a=sendrecv.

##
T 13.x.y.z:44557 -> a.b.c.189:5060 [AP]
SIP/2.0 180 Ringing.
CSeq: 102 INVITE.
Call-ID: 5e67085670baa1be07d898c605240dfd@a.b.c.189:5070.
From: <sip:0878209509@a.b.c.189:5070>;tag=as101814de.
To: <sip:9999990@a.b.c.189:5060>;tag=41f886b6.
Via: SIP/2.0/TCP a.b.c.189;branch=z9hG4bKa4d7.55e5aded836f1ad498a6289d5083fa44.0;rport=5060,SIP/2.0/UDP a.b.c.189:5070;received=a.b.c.189;branch=z9hG4bK05a2b5f0;rport=5070.
Record-Route: <sip:a.b.c.189;transport=tcp;r2=on;lr=on>,<sip:a.b.c.189;r2=on;lr=on>.
Contact: "9999990" <sip:9999990@172.17.0.2:44557;transport=tcp;registering_acc=vc2_xxxxx_com>.
User-Agent: Jigasi1.1.SNAPSHOTLinux.
Content-Length: 0.
.

##
T 13.x.y.z:44557 -> a.b.c.189:5060 [AP]
SIP/2.0 486 Busy here.
CSeq: 102 INVITE.
Call-ID: 5e67085670baa1be07d898c605240dfd@a.b.c.189:5070.
From: <sip:0878209509@a.b.c.189:5070>;tag=as101814de.
To: <sip:9999990@a.b.c.189:5060>;tag=41f886b6.
Via: SIP/2.0/TCP a.b.c.189;branch=z9hG4bKa4d7.55e5aded836f1ad498a6289d5083fa44.0;rport=5060,SIP/2.0/UDP a.b.c.189:5070;received=a.b.c.189;branch=z9hG4bK05a2b5f0;rport=5070.
Contact: "9999990" <sip:9999990@172.17.0.2:44557;transport=tcp;registering_acc=vc2_xxxxx_com>.
User-Agent: Jigasi1.1.SNAPSHOTLinux.
Content-Length: 0.
.

##
T a.b.c.189:5060 -> 13.x.y.z:44557 [AP]
ACK sip:9999990@172.17.0.2:44557;transport=tcp;registering_acc=vc2_xxxxx

Its over TCP.

There is nothing captured in /var/log/jitsi - @damenecho on another post you said there should be pcaps in there.

Hi all,

I discovered that docket-jitsi-meet changes the name of the Jitsi-Conference-Room to X-Room-Name.

Having changed my SIP server to make an X-Room-Name I’m getting further (though still not working).

I’ll post a new post with the next issue.

Thanks,
Steve