[asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

[Marek Červenka]

my experience with pjsip for webrtc
http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html


Dne 18.2.2016 v 15:36 Olivier napsal(a):
>
>
> 2016-02-18 14:57 GMT+01:00 Simon Hohberg 
> <simon.hohberg at mcs-datalabs.com <mailto:simon.hohberg at mcs-datalabs.com>>:
>
>
>         Is it implied here that both HTTPS and WSS must also come from
>         the same server (Same Origin Policy) ?
>
>     No, the same origin policy does not apply to web sockets.
>
>         Then, can I also install my own WebRTC demo page on my own
>         private  Asterisk server and access this demo page through HTTPS ?
>         If I'm not mistaken, this should fulfill all requirements.
>
>     It doesn't matter where the asterisk server is hosted. It is
>     important where the web application comes from. If you don't want
>     to use https and wss you only have the option to host the web app
>     locally (on the same machine as the browser that loads the page),
>     which probably makes sense only for development. Otherwise you
>     have to use https and wss for the reasons discussed earlier.
>
>     Hope it helps.
>
>
>
> At least, it helped me to realize I still have several more things to 
> learn ;-)
>
> My setup is the following:
> - an asterisk server,
> - a PC,
> - asterisk server and PC are installed on the same LAN
> - sipM5 live demo outside my LAN
> - no NAT/PAT configuration allowing incoming communications from the 
> outside.
>
> Is using sipML live demo as a way to rapidly test private Asterisk 
> WebRTC capabilies, something achievable ?
> What would keep this from working ?
>
>


-- 
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Marek Cervenka
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