WebRTC,STUN/TURN无法在局域网外工作

Question

index.html(Offerer)

var socket = io.connect('http://127.0.0.1:80'); //socket.io
socket.emit("player 1");

var iceServers = {
    iceServers: [
        {"url":"stun:turn1.xirsys.com"},
        {"username":"myusername","url":"turn:turn1.xirsys.com:443?transport=udp","credential":"mycredential"},
        {"username":"myusername","url":"turn:turn1.xirsys.com:443?transport=tcp","credential":"mycredential"}
    ]
};

var offererDataChannel, answererDataChannel;

var Offerer = {
    createOffer: function () {
        var peer = new PeerConnection(iceServers);
        var dataChannelOptions = {
            reliable: true,
            ordered: false
        };
        offererDataChannel = peer.createDataChannel('channel', dataChannelOptions);
        setChannelEvents(offererDataChannel);
        peer.onicecandidate = function (event) {
            if (event.candidate) {
            socket.emit("candidate", event.candidate);
        }
        };
        peer.createOffer(function (sdp) {
            peer.setLocalDescription(sdp);
            socket.emit("sdp", sdp);
        }, function (err) { peer.close(); });
        this.peer = peer;
        return this;
    },
    setRemoteDescription: function (sdp) {
        this.peer.setRemoteDescription(new SessionDescription(sdp));
    },
    addIceCandidate: function (candidate) {
        this.peer.addIceCandidate(new IceCandidate({
            sdpMLineIndex: candidate.sdpMLineIndex,
            candidate: candidate.candidate
        }));
    }
};

var Answerer = {
    createAnswer: function (offerSDP) {
        var peer = new PeerConnection(iceServers);
        peer.ondatachannel = function (event) {
            answererDataChannel = event.channel;
            setChannelEvents(answererDataChannel);
        };
        peer.onicecandidate = function (event) {
            if (event.candidate) {
                socket.emit("candidate", event.candidate);
            }
        };
        peer.setRemoteDescription(new SessionDescription(offerSDP));
        peer.createAnswer(function (sdp) {
            peer.setLocalDescription(sdp);
            socket.emit("sdp", sdp);
        }, function (err) { peer.close(); });
        this.peer = peer;
        return this;
    },
    addIceCandidate: function (candidate) {
        this.peer.addIceCandidate(new IceCandidate({
            sdpMLineIndex: candidate.sdpMLineIndex,
            candidate: candidate.candidate
        }));
    }
};

var peer = null;

socket.on("client is connected", function () {
    peer = Offerer.createOffer();
    socket.on("candidate", function (candidate) {
        peer.addIceCandidate(candidate);
    });
    socket.on("sdp", function (sdp) {
        peer.setRemoteDescription(sdp);
    });
});

我有另一个文件stream.html,它为回答者做了类似的事情.

在setChannelEvents功能上有信道onmessage的实现.

这在我家的局域网上完美运行.

我已经使用ngrok服务将其放在网上,以便在我的本地网络之外进行尝试,并且无法正常工作.

我正在使用谷歌浏览器(更新到最新版本,目前为35.0.1916.153).

有什么东西我不见了吗？WebTTC有TURN的工作示例我可以试试吗？

如果需要,我可以加入剩余的代码加的一些输出candidate和sdp.

注意:当我启动ngrok时,套接字上的IP被修改.

Answer 1

XirSys的家伙在这里.[=

我不太确定你的错误是什么,除了它不起作用.如果错误只是视频不流动,您应该知道TURN将无法正常工作,因为您已经嵌入了TURN的凭据,而这些凭据现在已经过期了.使用XirSys时,您必须致电/getIceServers获取与您的帐户关联的"新鲜"STUN和TURN服务器.每次启动呼叫时,都必须发出此POST请求并将结果放入iceServers变量中.

为了快速了解我们的平台,我建议您阅读以下指南:

• 介绍
• 快速入门指南
• 我们还有一些易于遵循的指南,可以将其他WebRTC API(包括SimpleWebRTC和EasyRTC)挂接到我们的STUN和TURN服务器中.

非常感谢您对我们的服务表示出兴趣,如果您有任何问题或意见,请与我们联系.