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<tt>Hi all,<br>
<br>
I guess you all noticed that a important piece of the OpenSIPS 3.1
is the Call API. Still, will haven't shared so much information on
that, so let me bring some light here (or an update on the topic).<br>
<br>
The calling API is offered by a new separate software (external to
OpenSIPS) called "<a moz-do-not-send="true"
href="https://github.com/OpenSIPS/call-api">Call API</a>". And
this Call API uses OpenSIPS as a SIP stack in order to run the
calls.<br>
<br>
So, the Call API engine seats between the actual API user and
OpenSIPS, acting as an enabler between the two sides.<br>
<br>
On the user side, the Call API:<br>
* provides an WebSockets based API <br>
* offers commands </tt><tt>start, terminate, mute/unmute and
transfer the calls hosted on OpenSIPS<br>
* feeds back the user with events about the manged calls.<br>
<br>
On the OpenSIPS side, the Call API:<br>
* talks to OpenSIPS via the MI interface (MI datagram)<br>
* subscribes for events via the event interface<br>
* uses the new "callops" module for a better grip and control
over the calls in OpenSIPS<br>
<br>
<br>
The OpenSIPS side was completed, as part of the OpenSIPS 3.1
release, but we are still working on the actual Call API to
complete some logic on managing the calls and reporting events.<br>
<br>
We expect the have this work completed in the next 2 weeks, with
full documentation, usage examples/scenarios and blog posting. And
of course with a first release of the Call API :)<br>
<br>
<br>
Best regards,<br>
</tt>
<pre class="moz-signature" cols="72">--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="https://www.opensips-solutions.com">https://www.opensips-solutions.com</a>
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