One way audio on conference

Hi,

This is a followup to my post “one way garbled audio”. I am trying to simplify things a bit and also there does not seem to be a problem with garbled audio.

I have the following setup:

  • One FreePBX Box behind Firewall (NAT), with ports 80/TCP, 5060/UDP and 10000:20000/UDP open.

  • One PJSIP Trunk to my SIP provider sip.1und1.de

  • One inbound route leading to one conference room secured with PIN

  • Two PJSIP extensions, 100 and 200, unused in this scenario

  • Two Smartphones
    Phone1: Nokia 5, Vodafone
    Phone2: iPhone 7, Telekom (Germany)
    calling in via mobile network (LTE) dialling the 1und1 number registered via the trunk.

What happens:

Phone1: Dials in, I hear IVR asking for PIN and name. After answering the questions, I am in the conference alone.

Phone2: Dials in, I hear absolutely nothing. By watching log file I know when which question is asked, answer the questions and join the conference, as announced on phone1. Phone1 can hear me but not vice versa.

Reversing the calls (Phone2 first) makes no difference. Phone2 is always deaf.

The only difference between the phones that I deem relevant here, are the different providers. Other than that, all communication goes over the same channels.

Changing the scenario:

  • Inbound route points to new extension 300
  • Third Phone3 connects to ext 300 using Acrobits Groundwire
  • Either Phone1 or Phone2 dialling in via mobile network can talk to Phone3 fine!

Isn’t that strange?

-Heinrich

I recall having a similar problem with Jigasi. People joining a Jitsi conference via mobile phone could not hear anything. Someone posted a workaround by adding

org.jitsi.impl.neomedia.transform.csrc.CsrcTransformEngine.DISCARD_CONTRIBUTING_SOURCES=true

to sip-communicator.properties

Does this ring any bells? Otherwise I really don’t know what to look for

-Heinrich