Hello,

I’m using Asterisk 13.9.1 (compiled from source) in Ubuntu 14.04.4.

I’m testing webrtc with jscommunicator (www.jscommunicator.org), and I’m using wss. My problem is as follows:

If I set

dtls_cipher=ALL

I can get audio with Firefox 47.0.1, but with Chrome 52.0.2743.60 beta, I just get log entry in messages.log telling about SSLv3 handshake failure; I’m not sure, but I suppose that I might get similar results with other Chrome versions.

If I set

dtls_cipher=ALL:!SSLv3

connection is established with both browsers, but there is no audio.

At the moment I have no idea what to try next.

Best,

Teijo