To verify WebRTC support follow these steps:
- From your Chrome browser, go to https://apprtc.appspot.com. If you are able to see a self-view, this indicates that Chrome can access your camera and microphone, and also indicates that STUN packets are not being blocked on the part of the network visiting that site and that the capabilities of the device will cope.
- Copy the link shown at the bottom and send it to someone and initiate a direct (point-to-point) call with live two-way audio and video.
Success of these tests indicates that WebRTC calls with the Cisco Meeting Server should also be successful. If you have the same firewall configurations between the Meeting Server and the browser as you do between the two browsers in the call.
Should any of these tests fail, computer or network issues should be investigated.
Note: This site does not provide much feedback. However, Chrome WebRTC implementers are adding ways to retrieve information such as packet loss and bandwidth estimates. When this happens, we will be able to display these statistics so that you don’t have to go to chrome://webrtc-internals. Timescales for this feature are currently unavailable.