We are using WebRTC via Doubango’s "webrtc2sip" with Asterisk for 2-way audio within browsers (no video necessary). We are very successful using this very same architecture within Chrome, Opera and Safari. However, when using Firefox 4x + (specifically testing with Firefox 47.x and 48.x), we are not able to establish consistent bi-direction audio connectivity. By consistent, this means that we DO establish audio, one out of every 30 to 50 attempts. By doing SIP traces and TCP / UDP dumps, we have been able to determine that the reason for this inconsistency (really it's more like non-functioning), is because Firefox does not send out the DTLS "Client Hello". This messaging is consistent in other browsers. As mentioned before, only with Firefox does it seem to basically almost never work. I guess my question would be: >>> Is if there is a way to force Firefox to send that DTLS Handshake, "Client Hello", so that audio can be established in a more consistent fashion? Thanks to all in advance. _____________________ (Attached are images of SIP traces / packet dumps of what what a successful audio stream connection looks like in Firefox [or any other browser] compared to an unsuccessful attempt):