Jigasi dial-in user can't be heard till any web participant unmute himself/herself

Hey guys, I’m seeing this weird issue where dial-in user is able to join the conference, however web and dial-in user can’t hear each other if both are muted. As soon as web user unmutes, both are able to hear each other and continue the conference.

And, then even if web user mutes/unmutes, everything still works as expected.

In short: web user needs to be unmuted or unmute them for the first time when a dial-in user joins the call.

Any idea, what might be the issue here? Thanks!

What is the sip side? What codec are you using there?
In general there is a hole punch packets that are sent to avoid this. In the past I have seen some providers to ignore it which can cause this.

@damencho We are having our own sip. Regardless, it has been working fine for us till now. As soon as, we upgraded our jitsi (stable-7882) and moved to Kubernetes - we have started seeing this issue. Not sure, if something has been added as part of Jigasi upgrade recently.

Any way to figure out issue and fix this?

Check whether in this case the 3 udp packets for hole punching are arriving to the sip side.

You can try setting:
net.java.sip.communicator.impl.protocol.HOLE_PUNCH_PKT_COUNT=16
to increase the number of packets and see whether that changes anything …