My main reason not to upgrade to Ast 13 is because I'm afraid of losing
functionality as there are certain functions deprecated/replaced. This
can also cause headache :-)
I will do so if there is no other option.
But still, I don't see why Ast 13 would differ so much in this case ? If
ICE and NAT is working (not causing problems) why should Ast 13 bring me
audio and Ast 12 don't ??
I indeed use SIPML5 demo as quick test-case. So do many tutorials on the
web.
Self-signed certificates should be OK as long as they are imported in
the browser. Never knew this could cause audio problems ?
Kind regards.
On 11-08-16 16:25, Jonathan H wrote:
I'm genuinely fascinated why you are insisting on using a version of
Asterisk almost 3 years old, for which EOL support ended last year.
Is there any particular reason you cannot or will not use the current
version as others have suggested?
Also, I see you are using Doubango and WebRTC, but in the logs, I see
WS and WSS.
You NEED to be using 100% WSS otherwise you've not got a hope in hell
of anything working with WEBRTC.
Check the console of the web browser you are trying to make the call
from (CTRL-SHIFT-I in Chrome on Windows, for example).
Also, you'll need to be using valid certificates - self-signed
certificates won't work for any current implementation of WebRTC that
I know of, certainly not if anything involves current versions of
Chrome or Firefox. That said, LetsEncrypt certs work fine for this, so
no need to spend out on one.
Switch to Asterisk 13.10 and save yourself a whole lotta headache.
On 11 August 2016 at 15:09, Jonas Kellens <[email protected]
<mailto:[email protected]>> wrote:
Hello
Using Asterisk 12.8.2.
On 10-08-16 22:03, Matt Fredrickson wrote:
My suggestion is to verify and debug against Asterisk 13
first, and
then you can try backing down versions, rather than reverse.
WebRTC
is a rapidly moving target, and has required ongoing changes
that may
not have made it into older and feature frozen versions of
Asterisk.
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