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

[Olivier]

2016-02-19 12:01 GMT+01:00 Marek Červenka <cervajs at fpf.slu.cz>:

> on my own server
>

Today, I'm back from holidays trip.

First of all, thanks for replying !

I'll try to use jssip as you suggested.

Anyway, I'm still failing to understand if wiki's page [1] is still valid
with Asterisk 13, and if it's not valid anymore, which is the main change
that prevent things to work.

[1] https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5


>
> 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>:
>
>> 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>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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