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<tt>Hi Alex,<br>
<br>
To make a kind of WS<>UDP gateway you need a complete rework
of the script presented in the tutorial, as it is a completely
different SIP scenario. Not sure what are your SIP/OpenSIPS skills.<br>
<br>
But, there is a simpler alternative . Instead of a GW, you can
make OpenSIPS as a sub-server for the WS extensions:<br>
<br>
Registration handling:<br>
<br>
1) WS extensions register only with OpenSIPS (as right now) -
authentication is done by OpenSIPS<br>
2) OpenSIPS registers the 3 extensions into </tt><tt>OmniPCX
using the uac_registrar <br>
<br>
By this, we simply add the uac_registration and you achieve kind
of decoupled 2 steps registration (with a minimum change in the
cfg)<br>
<br>
<br>
Inbound calls:<br>
<br>
1) </tt><tt><tt>OmniPCX will send all the calls (from other
extensions) for the WS extension to OpenSIPS (due the
registration via uac_registrar) - this is default behavior , so
nothing to change<br>
2) In OpenSIPS, when receiving calls, you need to authorize (by
IP) the calls from </tt></tt><tt><tt><tt><tt>OmniPCX - and as
the current script does, you will handle them via the local
opensips usrloc -> calls are sent to WS extension<br>
<br>
</tt></tt></tt></tt><br>
<tt><tt><tt><tt><tt>Outbound calls:<br>
<br>
1) when you receive a call from a WS extension, you have
to check if the call is for a local extension (on
opensips) or for an extension in </tt></tt></tt></tt></tt><tt><tt><tt><tt><tt><tt>OmniPCX<br>
2) if call is local (WS to WS) you will do authentication
for the call<br>
3) if the call is to be sent to </tt></tt></tt></tt></tt></tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt>OmniPCX,
simply send the call to </tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt>OmniPCX
without auth - the auth will be
done by </tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt>OmniPCX
as for any other
extension<br>
<br>
<br>
Hopefully this will
work for you :)<br>
<br>
</tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt>Best
regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a>
OpenSIPS Bootcamp 2017, Houston, US
<a class="moz-txt-link-freetext" href="http://opensips.org/training/OpenSIPS_Bootcamp_2017.html">http://opensips.org/training/OpenSIPS_Bootcamp_2017.html</a>
</pre>
<div class="moz-cite-prefix">On 06/29/2017 11:54 AM, Alex
Megalokonomos wrote:<br>
</div>
<blockquote
cite="mid:CALwx21oOpEXVPn3rqiY36+XP-S1QwqyyY+XZJMvZXfM5n7Zv1Q@mail.gmail.com"
type="cite">
<div dir="ltr"><img moz-do-not-send="true" class="mailtrack-img"
style="float:right;" alt=""
src="https://mailtrack.io/trace/mail/1676647efd99eeceab44e5dc440be7d1e78a150f.png?u=1422671"
height="0" width="0">Hello Bogdan,
<div><br>
</div>
<div>Yes, a gateway from WS to UDP (as well as DTLS-SRTP to RTP
in order for it to work) is exactly what we're looking for.</div>
<div><br>
</div>
<div>Unfortunately our Alcatel OmniPCX call center is a
proprietary system that only allows for a limited number of
SIP extensions (served from what appears to be an outdated
customised Kamailio 3.2.2 from what I can tell from the
headers.</div>
<div><br>
</div>
<div>For our normal internal office use it all works fine.</div>
<div><br>
</div>
<div>However we have 3 customer support lines that are currently
routed to 3 extensions via OmniPCX.</div>
<div><br>
</div>
<div>We want to integrate these to our custom web-based CRM and
the best way for us to do it is to use something like SIP js
to handle and log calls, identify calling parties, bring up
customer details etc.</div>
<div><br>
</div>
<div>Since the kamailio version inside OmniPCX does not support
ws/webrtc we are looking to set up Opensips in exactly the way
you described as a gateway/proxy for everything in order to
convert the UDP-only sip extensions to ws+ webRTC capable
ones.</div>
<div><br>
</div>
<div>I have used this tutorial <a moz-do-not-send="true"
href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1">http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1</a>
to get what I assume is half the work (for RTP proxying) but
I havent figured out the rest yet.</div>
<div><br>
</div>
<div>Best regards,</div>
<div>Alex</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Thu, Jun 29, 2017 at 11:43 AM,
Bogdan-Andrei Iancu <span dir="ltr"><<a
moz-do-not-send="true" href="mailto:bogdan@opensips.org"
target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:bogdan@opensips.org">bogdan@opensips.org</a></a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> <tt>Hi Alex,<br>
<br>
First, some questions regarding the desired topology:<br>
1) the WS end-points should register in OpenSIPS or
all the way into Kamailio ?<br>
2) also, the calls from the WS end-points should be
all the time sent to Kamailio ?<br>
<br>
More or less, what I'm asking is : is OpenSIPS suppose
to act as a gateway from WS to UDP , but pass all the
resulting traffic to Kamailio ?<br>
<br>
Regards,<br>
</tt>
<pre class="m_-6874691586421181616moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" class="m_-6874691586421181616moz-txt-link-freetext" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<wbr>com</a>
OpenSIPS Bootcamp 2017, Houston, US
<a moz-do-not-send="true" class="m_-6874691586421181616moz-txt-link-freetext" href="http://opensips.org/training/OpenSIPS_Bootcamp_2017.html" target="_blank">http://opensips.org/training/<wbr>OpenSIPS_Bootcamp_2017.html</a>
</pre><div><div class="h5">
<div class="m_-6874691586421181616moz-cite-prefix">On 06/28/2017 12:47 PM, Alex
Megalokonomos wrote:
</div>
</div></div><blockquote type="cite"><div><div class="h5">
<div dir="ltr"><img moz-do-not-send="true" class="m_-6874691586421181616mailtrack-img" style="float:right" alt="" src="data:image/gif;base64,R0lGODlhAQABAIAAAAAAAP///yH5BAEAAAAALAAAAAABAAEAAAIBRAA7" height="0" width="0">Hello,
<div>
</div>
<div>We have the following scenario: our office call center is
an Alcatel OmniPCX Office setup.</div>
<div>
</div>
<div>This handles most of our needs and also provides 4 SIP
extensions.</div>
<div>
</div>
<div>These are provided by what appears to be a Kamailio SIP
server v 3.2.2 (no webrtc or websockets support)</div>
<div>
</div>
<div>What we would like to do is set up an OpenSIPS instance to
handle WebRTC and proxy everything to this Kamailio SIP
server.</div>
<div>
</div>
<div>The idea is to allow a web client (using sip js or
something similar) to register / make / receive calls as one
of the Kamailio extensions.</div>
<div>
</div>
<div>
<div>I think half of the configuration is this : <a moz-do-not-send="true" class="m_-6874691586421181616moz-txt-link-freetext" href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1" target="_blank">http://www.opensips.org/<wbr>Documentation/Tutorials-<wbr>WebSocket-2-1</a></div>
<div>
</div>
<div>which I've already completed and indeed, clients can
register to opensips and chat/make calls over websockets
between them.</div>
<div>
</div>
<div>How do I go about proxying registrations/invites/etc to
the kamailio server instead?</div>
<div>
</div>
<div>best regards</div>
</div>
</div>
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</pre>
</span></blockquote>
</div>
</blockquote></div>
</div>
</blockquote>
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