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
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