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<div class="moz-cite-prefix">my experience with pjsip for webrtc<br>
<a class="moz-txt-link-freetext" href="http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html">http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html</a><br>
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Dne 18.2.2016 v 15:36 Olivier napsal(a):<br>
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<blockquote
cite="mid:CAPeT9jgo-EVan_TeuR-Ei=556VsDaubVMUi-SiMK611i6FMY6w@mail.gmail.com"
type="cite">
<div dir="ltr"><br>
<div class="gmail_extra"><br>
<div class="gmail_quote">2016-02-18 14:57 GMT+01:00 Simon
Hohberg <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:simon.hohberg@mcs-datalabs.com"
target="_blank">simon.hohberg@mcs-datalabs.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left:1px solid
rgb(204,204,204);padding-left:1ex"><span class=""><br>
<blockquote class="gmail_quote" style="margin:0px 0px
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Is it implied here that both HTTPS and WSS must also
come from the same server (Same Origin Policy) ?<br>
</blockquote>
</span>
No, the same origin policy does not apply to web sockets.<span
class=""><br>
<br>
<blockquote class="gmail_quote" style="margin:0px 0px
0px 0.8ex;border-left:1px solid
rgb(204,204,204);padding-left:1ex">
Then, can I also install my own WebRTC demo page on my
own private Asterisk server and access this demo page
through HTTPS ?<br>
If I'm not mistaken, this should fulfill all
requirements.<br>
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</span>
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.<br>
<br>
Hope it helps.</blockquote>
<div><br>
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<div>At least, it helped me to realize I still have several
more things to learn ;-)<br>
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<div>My setup is the following:<br>
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<div>- an asterisk server,<br>
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<div>- a PC,<br>
</div>
<div>- asterisk server and PC are installed on the same LAN<br>
</div>
<div>- sipM5 live demo outside my LAN<br>
</div>
<div>- no NAT/PAT configuration allowing incoming
communications from the outside.<br>
</div>
<div><br>
Is using sipML live demo as a way to rapidly test private
Asterisk WebRTC capabilies, something achievable ?<br>
</div>
<div>What would keep this from working ?<br>
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<br>
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<br>
</blockquote>
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<pre class="moz-signature" cols="72">--
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Marek Cervenka
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