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

[Olivier]

2016-02-18 15:42 GMT+01:00 Marek Červenka <cervajs at fpf.slu.cz>:

> my experience with pjsip for webrtc
> http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html
>
>
> Yes I saw this post earlier today.
Having to fight 14 days scared me a bit !

Did you set sipml5 on your own server or did you use Live demo (
https://www.doubango.org/sipml5/call.htm?svn=241) ?



> 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>:
>
>>
>> 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 ?
>
>
>
>
> --
> ---------------------------------------
> Marek Cervenka
> =======================================
>
>
> --
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