Does not change, it was taken into account already.
I don't understand your point regarding translator/relay mode and multiple SSRC.
if the sip extension I call is on a sipserver (asterisk) which does not allow direct media then the RTP flows
is between Ofmeet <=> Asterisk and then Asterisk <=> Sip client.
this is a better way to handle codec transcoding when both do not use the same ones.
I will try with a sip client registered to the embedded ofmeet sipserver to see if this is better.