WebRTC is the newest of the communications standards growing at W3C, providing web developers with APIs to obtain access to local cameras and microphones and send that audio and video directly to any other browser, without plug-ins. The media flows peer-to-peer, without requiring a media gateway or relay in many cases. This course gives attendees a solid introduction into the standard that could soon become the foundation for all live person-to-person communication. The tutorial covers the VoIP technologies underlying WebRTC, common deployment models, and the APIs for using the standard. See demos, walkthroughs of real code, and—time permitting—some live coding.