https://fedrtc.org has been running for a while now and this has given many people a chance to get a taste of regular SIP and WebRTC-based SIP. As suggested in Zoltan's blog, it has convenient integration with Fedora SSO and as the source code is available, people are welcome to see how it was built and use it for other projects.
If you tried any of FedRTC.org, rtc.debian.org or meet.jit.si using Chrome/Chromium on Linux, you may have found that the call appears to be connected but there is no media. This is a bug and the Chromium developers are on to it. You can work around this by trying an older version of Chromium (it still works with v37 from Debian wheezy) or Firefox/Iceweasel.
WebRTC offers many great possibilities for people to quickly build and deploy RTC services to a large user base, especially when using components like JSCommunicator or the DruCall WebRTC plugin for Drupal.
Native applications and mobile apps like Lumicall continue to offer the most optimized solution for each platform although WebRTC currently offers the most convenient way for people to place a Call me link on their web site or portal.
The RTC Quick Start Guide offers step-by-step instructions and a thorough discussion of the architecture for people to start deploying RTC and WebRTC on their own servers using standard packages on many of the most popular Linux distributions, including Debian, Ubuntu, RHEL, CentOS and Fedora.