One-way audio with SIP calls

Dear Jitsi community,

I’m encountering difficulties while trying to setup jigasi with my SIP provider (OVH).

I installed jitsi on a Debian LXC container using jitsi’s download.jitsi.org repository.

What works:

  • internal hash auth
  • room creation by web users
  • audio/video between web users
  • incoming calls join default room
  • outgoing calls to invite people to a room
  • web users can hear sip users

What doesn’t work:

  • sip users can’t hear web users

I don’t really know what to do now.

Here are the warnings in jigasi.log (on jigasi start)

2022-10-06 02:01:37.252 GRAVE: [15] DeviceConfiguration.registerCustomRenderers#1056: Failed to register custom Renderer org.jitsi.impl.neomedia.jmfext.media.renderer.audio.PulseAudioRenderer with JMF.
2022-10-06 02:01:43.881 AVERTISSEMENT: [15] CallControl.<init>#122: Always trust in remote TLS certificates mode is enabled
2022-10-06 02:01:48.182 AVERTISSEMENT: [35] net.java.sip.communicator.impl.resources.ResourceManagementServiceImpl.getSettingsInt: Missing resource for key: net.java.sip.communicator.SIP_PREFERRED_CLEAR_PORT
2022-10-06 02:01:48.403 AVERTISSEMENT: [35] net.java.sip.communicator.impl.resources.ResourceManagementServiceImpl.getSettingsInt: Missing resource for key: net.java.sip.communicator.SIP_PREFERRED_SECURE_PORT
2022-10-06 02:01:49.635 AVERTISSEMENT: [45] SipHealthPeriodicChecker.create#169: No health check started, no HEALTH_CHECK_SIP_URI prop.

Here are the warnings in jigasi.log (on sip call)

2022-10-06 12:41:59.908 GRAVE: [294] net.sf.fmj.media.Log.error:   Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
2022-10-06 12:41:59.908 GRAVE: [294] net.sf.fmj.media.Log.error: Failed to prefetch: net.sf.fmj.media.ProcessEngine@770ae8b0
2022-10-06 12:41:59.909 GRAVE: [293] net.sf.fmj.media.Log.error: Error: Unable to prefetch net.sf.fmj.media.ProcessEngine@770ae8b0

Thank you for your help :slight_smile:

Does that provider does latching or it depends on stun config in the client

I don’t know that, so I’m going to ask them.

Hi,
I wasn’t able to get more info about this latching thing.
However, I was able to fix the problem by adding the following configuration line in jigasi’s sip-communicator.properties:
org.jitsi.impl.neomedia.transform.csrc.CsrcTransformEngine.DISCARD_CONTRIBUTING_SOURCES=true

Though, I don’t know what it does and why it works. I just found this from Jitsi + Jigasi, one way audio - #3 by TPdev

Thanks @damencho for your help with this problem.

Hum maybe we need to add that as default … Will think about it.

This means that the remote side(sip) is not handling rtp correctly and is getting messed up when there are rtp header extensions.

1 Like

You may report it back to them so they at least know it.