Most people don't have time to read manuals. The Debian SIP service has a reasonably thorough user guide with screenshots but some people may just want to have this level of detail if they really have to troubleshoot a problem.
Just set up your db.debian.org RTC password and use it to log in to the https://rtc.debian.org site.
It all runs in the browser, no plugin or configuration required.
The current range of WebRTC browsers seem to work well for consumer and small office networks with NAT routers or mobile/cell networks. Corporate networks with an outright block on UDP will be supported soon.
The Chromium 31 package on wheezy appears to work fine and was used in some of the demos at DebConf13
For Iceweasel/Firefox users, it is not possible to use the version 17 package from wheezy at all. At the time of writing, Iceweasel 24 is in jessie and sid while version 27 is available upstream. Mozilla gradually introduced WebRTC support on both desktop and Android between versions 20 to 24 and it is still evolving.
If you don't want to use Chromium and don't want to upgrade your desktop/laptop beyond wheezy, consider trying Firefox on an Android device.
There is no excuse for them to pester you with proprietary solutions any more.
They can dial your debian.org SIP address from https://freephonebox.net. You can even link to it from your web site or blog by adding the dial parameter to the URL, like https://freephonebox.net?dial=pocock%40debian.org
Bug reports and questions are always welcome. A good place for questions or complaints where the cause of the bug is not obvious is the Free-RTC mailing list.