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[Freeswitch-users] INCOMPATIBLE_DESTINATION sip softphone <-> jssip webrtc


 
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PostPosted: Fri Dec 11, 2020 10:59 am    Post subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION sip softphone &l Reply with quote

------ Start of attached email. Subject: INCOMPATIBLE_DESTINATION sip softphone <-> jssip webrtc ------
Hi FS users

I am stuck with INCOMPATIBLE reason of "Not Acceptable Here" 488 message.

Calling from sipphone to sipphone works, calling from webrtc to webrtc
works, calling a conference room from webrtc and sipphone (mixed) works,
but not when directly crossing over.

Sipphones tried are microsip and linphone

I use the "proxy_media=true" for the webrtc calls to work.

I have checked the codecs in the SIP bodies from both sides' INVITE
message, they seem to have common codecs in different orders (not sure
if this matters)

Websocket endpoint listens to 127.0.0.1 7443 behind apache ws_tunnel.
SIP 5060 listens on public IP

Not sure why either side won't cross over.

Text messages work fine from webrtc to sipphone.

Any ideas?





------ End of attached email ------
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