[asterisk-users] Asterisk, linphone, and jssip webrtc

[Henry S]

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