我正在查看webrtc.html和peerconnection_server演示，它在两个 Chrome 浏览器之间运行良好。我的问题是， webkitPeerConnection 的第一个参数到底是什么？
pc = new webkitPeerConnection("STUN stun.l.google.com:19302", onSignalingMessage);
它是 Google 为演示目的提供的第三方 STUN 服务器吗？如果在我的 JavaScript 代码中替换"stun.l.google.com:19302"为"toto"，我仍然可以进行视频通话。但由于我在同一个子网上，这可以解释......
STUN servers are used by both clients to determine their IP address as visible by the global Internet.If both the peers are behind the same NAT , STUN settings are not needed since they are anyways reachable form each other . STUN effectively comes into play when the peers are on different networks.
As we know that webRTC is peer to peer and the ice candidates are mandatory in webrtc. ICE functionality can be in any of the two ways , STUN and TURN .
There are many stun servers provided by google and other sites which one could use . You can also setup your own STUn server according to rfc5766.
Hope that gives a zest of what and how of stun .
如果两个客户端在 NAT 后面，则需要一个 stun 服务器才能使用 webrtc 进行通信。您将需要该 stun 服务器来确保 NAT 后面的人可以在您的网页上使用 webrtc 功能。