<div dir="ltr"><br><div class="gmail_extra"><br><div class="gmail_quote">2016-02-19 12:01 GMT+01:00 Marek Červenka <span dir="ltr"><<a href="mailto:cervajs@fpf.slu.cz" target="_blank">cervajs@fpf.slu.cz</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">
<div text="#000000" bgcolor="#FFFFFF">
<div>on my own server<br></div></div></blockquote><div><br>Today, I'm back from holidays trip.<br></div><div><br>First of all, thanks for replying !<br><br></div>I'll try to use jssip as you suggested.<br><br></div><div class="gmail_quote">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.<br><br>[1] <a href="https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5">https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5</a><br></div><div class="gmail_quote"><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div text="#000000" bgcolor="#FFFFFF"><div>
<br>
i want try jssip<br>
<a href="https://github.com/versatica/JsSIP" target="_blank">https://github.com/versatica/JsSIP</a><br>
it looks like a lot <span lang="en"><span>"livelier" than sipml5<br>
<br>
any experience with jssip?<br>
</span></span><br>
<br>
Dne 18.2.2016 v 16:01 Olivier napsal(a):<br>
</div><div><div class="h5">
<blockquote type="cite">
<div dir="ltr"><br>
<div class="gmail_extra"><br>
<div class="gmail_quote">2016-02-18 15:42 GMT+01:00 Marek
Červenka <span dir="ltr"><<a href="mailto:cervajs@fpf.slu.cz" target="_blank">cervajs@fpf.slu.cz</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">
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<div>my experience with pjsip for webrtc<br>
<a href="http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html" target="_blank">http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html</a><br>
<br>
<br>
</div>
</div>
</blockquote>
<div>Yes I saw this post earlier today.<br>
Having to fight 14 days scared me a bit !<br>
<br>
</div>
<div>Did you set sipml5 on your own server or did you use
Live demo (<a href="https://www.doubango.org/sipml5/call.htm?svn=241" target="_blank">https://www.doubango.org/sipml5/call.htm?svn=241</a>)
?<br>
</div>
<div><br>
</div>
<div> </div>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
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<div> Dne 18.2.2016 v 15:36 Olivier napsal(a):<br>
</div>
<span>
<blockquote 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 href="mailto:simon.hohberg@mcs-datalabs.com" target="_blank"></a><a 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><br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"> 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><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>
</blockquote>
</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>
<br>
</div>
<div>At least, it helped me to realize I still
have several more things to learn ;-)<br>
<br>
</div>
<div>My setup is the following:<br>
</div>
<div>- an asterisk server,<br>
</div>
<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>
</div>
</div>
<br>
</div>
</div>
</blockquote>
</span></div>
</blockquote>
</div>
</div>
</div>
</blockquote>
<br>
<pre cols="72">--
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Marek Cervenka
=======================================
</pre>
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