I has one problem, so i ask for help here.
I use asterisk to transfer webrtc to my sip gateway. but my sip gateway don’t support webrtc. so I use asterisk.
I find this work well, when i call one video or two video .
but when a call has set up. sip gateway add a new video (change one video to two video). the other one don’t receive invite.
Can i change asterisk configure to support this?
I find some community topic say asterisk is b2b model don’t tranfer reivite.
some body can tell me answer。 thinks。