Here are some sample pages that demonstrate basic WebRTC concepts. If you are new to WebRTC, you may want to check out this WebRTC overview first.
| getUserMedia Samples | |
| gum1.html | Shows how to access the webcam and display the local video in a <video/> element. | 
| gum2.html | Shows how to capture the current frame of video to a <canvas/>. | 
| gum3.html | Shows how to apply CSS filters to a <video/> and <canvas/> | 
| face.html | Shows how to perform face tracking using webcam video. | 
| local-audio-rendering.html | Shows usage of a local media stream connected to an HTML5 audio tag. | 
| local-audio-volume.html | Shows how to display the volume of a local audio track. | 
| PeerConnection Samples | |
| pc1-audio.html | Shows how to set up a simple 1:1 audio only call. | 
| pc1.html | Shows how to set up a simple 1:1 audio/video call. | 
| pc1_sdp_munge.html | Allows you to modify offer/answer sdp with pc1 demo. | 
| states.html | Shows RTCPeerStates and RTCIceConnectionStates in a simple 1:1 audio/video call. | 
| multiple.html | Shows how to set up multiple PeerConnections. | 
| constraints-and-stats.html | Shows how to pass constraints into the PeerConnection API, and query it for statistics. | 
| dtmf1.html | Shows how to send DTMF tones using PeerConnection API. | 
| dc1.html | Shows how to send Data using PeerConnection API. | 
| webaudio-and-webrtc.html | Captures and filters microphone input using WebAudio and sends it to a remote peer with an option to add an audio effect. | 
| create-offer.html | Shows the output of createOffer when various constraints are supplied. | 
| ice-servers.html | Tests gathering candidates from arbitrary STUN and TURN servers. |