Dear all, Thanks for the amazing product !
I’m trying my hands on a scenario involving Asterisk for video_sfu and a bunch of webrtc and sip clients. The condition is that every call must essentially be a conference call for the participants to add more dynamically. I front ended asterisk with Kamailio and RTPEngine for ssl offloading, load balancing, and registrations. So when one user is present in the conference (his invite SDP containing one audio and one video line) i get local video and MoH back, but when another user joins in and a re-invite is send from asterisk containing two video lines - one for the sendrecv and other for send only, rtpengine doesn’t process the sendonly video line - keeps it as is RTP/AVP instead of making it *UDP/TLS/RTP/SAVPF.* *Is there a way to achieve that? or I’m on the wrong path ?* BR, RM
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