DruCall is one of the easiest ways to get up and running with WebRTC voice and video calling on your own web site or blog. It is based on 100% open source and 100% open standards - no binary browser plugins and no lock-in to a specific service provider or vendor.
On Debian or Ubuntu, just running a command such as
# apt-get install -t jessie-backports drupal7-mod-drucall
The user interface
Most of my experience is in server-side development, including things like the powerful SIP over WebSocket implementation in the reSIProcate SIP proxy repro.
In creating DruCall, I have simply concentrated on those areas related to configuring and bringing up the WebSocket connection and creating the authentication tokens for the call.
Those things provide a firm foundation for the module, but it would be nice to improve the way it is presented and optimize the integration with other Drupal features. This is where the projects (both DruCall and JSCommunicator) would really benefit from feedback and contributions from people who know Drupal and web design in much more detail.
Benefits for collaboration
If anybody wants to collaborate on either or both of these projects, I'd be happy to offer access to a pre-configured SIP WebSocket server in my lab for more convenient testing. The DruCall source code is a Drupal.org hosted project and the JSCommunicator source code is on Github.
When you get to the stage where you want to run your own SIP WebSocket server as well then free community support can also be provided through the repro-user mailing list. The free, online RTC Quick Start Guide gives a very comprehensive overview of everything you need to do to run your own WebRTC SIP infrastructure.