Dial-in with more than three participants

Install & Config
#1

Hi!

I seem to have a variant of the common “more than two participants don’t work”-problem, but with a twist.

I have installed a Jitsi-instance on a Ubuntu-machine according to the documentation and also set up a SIP-account for Jigasi.

The following scenarios have been observed:

  • Web-user A creates a meeting
    User B dials in – quality is perfect
    User C dials in – user A can hear both, B and C hear cracking sounds

  • Web-user A creates a meeting
    User B connects via web – quality is perfect
    User C dials in – user A and B can hear everyone, C hears only cracking sounds

In summary: it seems that the return paths of all dial-in participants have a problem as soon as there are more than two participants in a meeting.

I have tried turning the translator (USE_TRANSLATOR_IN_CONFERENCE) on or off, but to no avail.

Best

#2

The sounds like you are doing multi streaming (translator is enabled) on a sip service that do not support that.
Does your sip service support translator mode?

#3

Thanks for getting back to me so quickly!
We a re using a (more or less) default Kamailio + rtpengine installation. I presume, that what Jigasi calls a “translator” is the “transcoder” there. In this case – yes, our rtpengine does support transcoding and yes, it supports both opus and the phone’s codec (PCMA).
I have, however, also tired the set-up with the translator disabled, and it didn’t work either…

#4

Nope translator, is the mode where jigasi just forwards any audio rtp stream it receives to the sip side. For a 3 way call where the third participant is jigasi it will forward two audio streams, two different ssrcs to the sip side, and the sip side needs to do the mixing.

#5

This explains why it doesn’t work with the translator enabled.
When the translator is disabled, Jigasi – or jvb – does the mixing. Any idea why it still doesn’t work?

#6

Jigasi does the mixing. maybe share your jigasi config, how is the cpu on the jigasi machine?