After changing the USE_TRANSLATOR_IN_CONFERENCE to true in
sip-communicator.properties file, Jigasi still works in mixer mode.
It looks like we also need to change the xmpp/webrtc side to work in
translator mode instead of mixer, however there is no such configuration
for xmpp/webrtc side.
I made the change in the service/protocol/Call.java as below marked by ^^^
From the wireshark capture between Jigasi and FreeSwitch, I can see Jigasi
sends two audio streams to FreeSwitch.
Damin/Emil, is my understanding correct, and what is the better way to set
XMPP/webrtc side to work in RTP translator mode?