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

[Marek Červenka]

on my own server

i want try jssip
https://github.com/versatica/JsSIP
it looks like a lot "livelier" than sipml5

any experience with jssip?


Dne 18.2.2016 v 16:01 Olivier napsal(a):
>
>
> 2016-02-18 15:42 GMT+01:00 Marek Červenka <cervajs at fpf.slu.cz 
> <mailto: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
>>     <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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