Hi All, I am new to VoIP world and trying to set up asterisk, linphone, and jssip webrtc. Settings: - transport_wss (127.0.0.1, apache ws_tunnel) - transport_tls (public ip port 5060) - use_avpf=yes - ice_support=yes - dtls enabled (letsencrypt) - rtcp_mux=yes - allow=vp8,g722,h263,h265,opus,ulaw Findings: - jssip webrtc <-> jssip webrtc (success video&audio, fail when dtls disabled) - linphone -> jssip webrtc (success video&audio) - jssip webrtc -> linphone (fail after 30 seconds, no video&audio) - linphone <-> linphone (fail, success when dtls disabled) Can anyone please help? Thanks