What is the USE_TRANSLATOR_IN_CONFERENCE parameter for?

Hello everyone,

I’m currently working on SIP integration with Jitsi.

There is USE_TRANSLATOR_IN_CONFERENCE in Jigasii configuration:

# can be enabled to disable audio mixing and use translator, jigasi will act as jvb, 
just forward every ssrc stream it receives.
#org.jitsi.jigasi.xmpp.acc.USE_TRANSLATOR_IN_CONFERENCE=true
  1. Can anyone clarify why the USE_TRANSLATOR_IN_CONFERENCE parameter is needed? For what purposes?

  2. When should I use it?

  3. What is the difference between modes when USE_TRANSLATOR_IN_CONFERENCE is set to true and false?
    How is the data flow changing?

  4. Does the statement “jigasi acts as jvb, just forward every ssrc stream it receives” mean
    that Jigasi can be used as a gateway for video calls?

  5. Is it possible to setup video call via Jigasi using USE_TRANSLATOR_IN_CONFERENCE?

Thanks.

Jigasi has two modes of operation when it comes to sip. The default one is to decrypt and audio decode the audio that comes from the bridge, normally opus and then mix it with rest of the streams and encodes it and send it to the sip side in the correct codec as one stream.
If the sip side supports opus and multiple streams, you can offload jigasi by using tranlslator mode. This is basically doing the same thing the videobridge is doing, routing packets. So whatever comes from the bridge is re-routed to the sip side as multiple streams from every participant.
We are using Voximplant for interconnecting meet.jit.si and jigasi to PSTN where multiple streams and mixing is supported and we are using this mode.

[quote=“Ivan_Morozoff, post:1, topic:55009”]
Does the statement “jigasi acts as jvb, just forward every ssrc stream it receives” [/quote]
Yes.

No. Jigasi is audio only.

Nope. For that we use the jibri approach, but it launches pjsua to establish the sip video call.

Thank you so much for your quick reply.