Pjsua starting error - Jibri - video SIP gateway

Requirement is to implement jitsi sip video gateway using jibri and Pjsip.
While initiating the call jibri is accepting the request but the Pjsua/Pjsip is not getting started. Need help to troubleshoot the same. @damencho @Aaron_K_van_Meerten

Error:

2020-10-07 08:22:43.334 INFO: [44] org.jitsi.jibri.util.JibriSubprocess.pjsua.launch() Starting pjsua with command pjsua --capture-dev=23 --playback-dev=24 --id “TestSip” sip:jibri@127.0.0.1 --config-file /home/jibri/pjsua.config --log-file /tmp/pjsua.out --max-calls=1 sip:11111@bjn.vc ([pjsua, --capture-dev=23, --playback-dev=24, --id, “TestSip” sip:jibri@127.0.0.1, --config-file, /home/jibri/pjsua.config, --log-file, /tmp/pjsua.out, --max-calls=1, sip:11111@bjn.vc])

2020-10-07 08:22:43.341 SEVERE: [44] org.jitsi.jibri.util.JibriSubprocess.pjsua.launch() Error starting pjsua

2020-10-07 08:22:43.344 INFO: [44] org.jitsi.jibri.service.impl.SipGatewayJibriService.onServiceStateChange() SIP gateway service transitioning from state Starting up to Running

Pjsua.config:

–capture-dev=7
–playback-dev=8
–video
–vcapture-dev=1
–no-color
–log-level=5
–app-log-level=5
–auto-update-nat=0
–disable-stun
–no-tcp
–dis-codec=GSM
–dis-codec=H263
–dis-codec=iLBC
–dis-codec=G722
–dis-codec=speex
–dis-codec=pcmu
–dis-codec=pcma
–dis-codec=opus
–add-codec=G722
–add-codec=opus
–no-vad
–ec-opt=2
–jb-max-size=30
–quality=10
–max-calls=1
–no-stderr
–log-file=/var/log/jitsi/jibri/pjsua.log
–auto-answer=200
–duration=7200
–stereo

Try launching it by hand with same params. You are using the pjsua from the jitsi repo, right? Use the branch jibri-2.10-dev1 or the jibri-2.9-dev2

1 Like

@damencho Thank you so much for your reply, have updated Jibri and Pjsua. It is working now.

But the user drops within few seconds from the conference, getting below message in Jibri log
SIP gateway service transitioning from state Running to Error: RemoteSipClientBusy SESSION Remote side busy
2020-10-11 22:11:33.932 INFO: [51] org.jitsi.jibri.api.xmpp.XmppApi.invoke() Current service had an error Error: RemoteSipClientBusy SESSION Remote side busy, sending error iq

Also, would like to know the sip Uri format for the dial-in option from cisco hardware to jitsi-meet.
eg: meetingId @ conference.domain-name/focus is this correct format? can you give some example? Thanks.