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
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