Tag : Jitsi sip client

WebRTC No Audio

Posted August 10, 2015 by Marek Cervenka & filed under Asterisk Users Comments: 7.

Tags: audio devices, chan_sip wss transport, chrome, Jitsi sip client

Im facing strange problem:Aasterisk13.5 + 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(chro..

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