Jigasi answers SIP caller but no audio, PulseAudioRenderer issue

I installed Jigasi with Jitsi on a Debian 10 Linode. Configured a SIP provider to connect to a default room, siptest. When I dial in the call connects but there is no audio. In /var/log/jitsi/jigasi.log I found the following problem with PulseAudioRenderer.

I have version pulseaudio/stable,now 12.2-4+deb10u1 amd64 of PulseAudio installed.
The same was reported in https://github.com/jitsi/jigasi/issues/43 but no one ever responded. Any suggestions would be appreciated.
Dennis

2020-11-26 22:32:23.253 SEVERE: [13] org.jitsi.impl.neomedia.device.DeviceConfiguration.log() Failed to register custom Renderer org.jitsi.impl.neomedia.jmfext.media.renderer.audio.PulseAudioRenderer with JMF.
java.lang.IllegalStateException: audioSystem
at org.jitsi.impl.neomedia.jmfext.media.renderer.audio.PulseAudioRenderer.(PulseAudioRenderer.java:156)
at org.jitsi.impl.neomedia.jmfext.media.renderer.audio.PulseAudioRenderer.(PulseAudioRenderer.java:136)
at java.base/jdk.internal.reflect.NativeConstructorAccessorImpl.newInstance0(Native Method)
at java.base/jdk.internal.reflect.NativeConstructorAccessorImpl.newInstance(NativeConstructorAccessorImpl.java:62)
at java.base/jdk.internal.reflect.DelegatingConstructorAccessorImpl.newInstance(DelegatingConstructorAccessorImpl.java:45)
at java.base/java.lang.reflect.Constructor.newInstance(Constructor.java:490)
at java.base/java.lang.Class.newInstance(Class.java:584)
at org.jitsi.impl.neomedia.device.DeviceConfiguration.registerCustomRenderers(DeviceConfiguration.java:1047)
at org.jitsi.impl.neomedia.device.DeviceConfiguration.(DeviceConfiguration.java:366)
at org.jitsi.impl.neomedia.MediaServiceImpl.(MediaServiceImpl.java:162)
at java.base/jdk.internal.reflect.NativeConstructorAccessorImpl.newInstance0(Native Method)
at java.base/jdk.internal.reflect.NativeConstructorAccessorImpl.newInstance(NativeConstructorAccessorImpl.java:62)
at java.base/jdk.internal.reflect.DelegatingConstructorAccessorImpl.newInstance(DelegatingConstructorAccessorImpl.java:45)
at java.base/java.lang.reflect.Constructor.newInstance(Constructor.java:490)
at java.base/java.lang.Class.newInstance(Class.java:584)
at org.jitsi.impl.libjitsi.LibJitsiImpl$ServiceLock.initializeService(LibJitsiImpl.java:196)
at org.jitsi.impl.libjitsi.LibJitsiImpl$ServiceLock.getService(LibJitsiImpl.java:131)
at org.jitsi.impl.libjitsi.LibJitsiImpl.getService(LibJitsiImpl.java:91)
at org.jitsi.impl.libjitsi.LibJitsiOSGiImpl.getService(LibJitsiOSGiImpl.java:95)
at org.jitsi.service.libjitsi.LibJitsi.invokeGetServiceOnImpl(LibJitsi.java:172)
at org.jitsi.service.libjitsi.LibJitsi.getMediaService(LibJitsi.java:124)
at net.java.sip.communicator.impl.neomedia.NeomediaActivator.start(NeomediaActivator.java:391)
at org.jitsi.impl.osgi.framework.BundleImpl.start(BundleImpl.java:307)
at org.jitsi.impl.osgi.framework.launch.FrameworkImpl.startLevelChanged(FrameworkImpl.java:472)
at org.jitsi.impl.osgi.framework.startlevel.FrameworkStartLevelImpl$Command.run(FrameworkStartLevelImpl.java:137)
at org.jitsi.impl.osgi.framework.AsyncExecutor.runInThread(AsyncExecutor.java:122)
at org.jitsi.impl.osgi.framework.AsyncExecutor.access$000(AsyncExecutor.java:28)
at org.jitsi.impl.osgi.framework.AsyncExecutor$1.run(AsyncExecutor.java:231)

That exception you ca ignore. Make sure your sip service is configured to do latching.

And check the codecs you use.

Thanks much for the response. I am new to much of this. I have been searching on the internet about latching. Do you have a link on how to configure a sip service to do latching or that at least gives some background? Is the sip service you refer to the sip service provider server or the server hosting jigasi?

What codec should I be looking for or do I investigate that any of the configured codecs are not actually installed?

Thanks much.

Latching should be configured in your sip service provider. In asterisk it used to be a setting nat=yes, for the codec again you should check what is your provider supporting …

I am investigating what to do with the sip service provider.

The provider web site says:

Supported Codecs

Codecs (G723/G729/G711)
The system is transparent for codec negotiations

In my jigasi sip-communication.properties the closest I could find was:

net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.G722/8000=700
Do I need to install one of G723,G729 or G711?

I think I need to integrate jigasi with something like Voximplant and there is a good guide to this on the Voximplant web site.

Can I also achieve the same with FreePBX instead? Does FreePBX provide the ability for a sip call to be routed to any room with a passcode (conference mapper) or directly with the room name in the url? Does it support outbound calls, mute/unmute functionality. Can FreePBX be installed on the same Linode(VM) as Jitisi Meet or does it require to be installed on another instance?
Thanks.

It can work with asterisk without a problem, in asterisk you can create IVR and set custom sip headers.
Should be fine and on the same machine

I installed this program and configured it, but I don’t hear anything or anyone, and others don’t hear me in the same way. Tell me which codecs should be installed and maybe there are some special settings?

Thanks much. That is the approach I am going to try. I noticed that there were entries s on how to integrate asterisk with jigasi in this forum but seem fragmented. Is there on link to the best tutorial on how to do this?