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. |