Hi all,
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).
The calling API is offered by a new separate software (external to
OpenSIPS) called "Call API <https://github.com/OpenSIPS/call-api>". And
this Call API uses OpenSIPS as a SIP stack in order to run the calls.
So, the Call API engine seats between the actual API user and OpenSIPS,
acting as an enabler between the two sides.
On the user side, the Call API:
* provides an WebSockets based API
* offers commands start, terminate, mute/unmute and transfer the calls
hosted on OpenSIPS
* feeds back the user with events about the manged calls.
On the OpenSIPS side, the Call API:
* talks to OpenSIPS via the MI interface (MI datagram)
* subscribes for events via the event interface
* uses the new "callops" module for a better grip and control over the
calls in OpenSIPS
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.
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 :)
Best regards,
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
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