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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