[asterisk-users] webrtc no audio

[Marek Cervenka]

hello,

i'm facing strange problem

asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230
person1 to person3 are behind different NATs
audio devices double checked

call from person1(chrome) to person2(chrome) works
call from person1(chrome) to person 3(chrome) - no audio on both side   
(RTP flowing only in one direction)
call from person2(chrome) to person 3(chrome) - no audio on both side   
(RTP flowing only in one direction)
BUT
call from person2(chrome) to person 3(Jitsi sip client) - works!

any tips howto find the problem?

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Marek Cervenka
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