Question

我的简单WebRTC javascript代码无法按预期工作。事实上，音频呼叫没有建立（请注意，我对WebRTC的了解最少，我通过查看互联网上的示例创建了这个）。该页面应在两个参与者之间发起音频呼叫。作为参与者之间的信令服务器，我使用了websocket-server。此服务器仅在参与者之间中继消息。在呼叫发起期间，消息实际上是通过websocket发送的（一个提议，几个候选人，一个答案和其他候选人）仍然，Firefox给了我“ICE失败，请参阅：webrtc了解更多细节”。两位参与者都落后于普通路由器我将尽快添加我的websocket-server的日志和about：webrtc（当然缩短）的示例。

为什么这段代码不起作用？我忽略了什么？

我的代码是（记住这只适用于firefox）：

ws = new WebSocket("ws://" + location.hostname + ":9000");
navigator.getUserMedia = function(a, b){ return navigator.mozGetUserMedia(a, b, error);};
offerOptions = {offerToRecieveAudio: 1, offerToRecieveVideo: 1};

var pc = new RTCPeerConnection({"iceServers": [
{url:'stun:stun.l.google.com:19302'},
{url:'stun:stunserver.org'},
]});
pc.onaddstream = function(obj) {
  if (obj.stream instanceof LocalMediaStream) return;
  var audio = document.createElement("audio");
  audio.controls = "true";
  audio.autoplay = "true";
  document.body.appendChild(audio);
  audio.srcObject = obj.stream;
}
pc.onicecandidate = function(evt){
        if (!evt.candidate) return;
        console.log(evt.candidate);
        ws.send(JSON.stringify(evt.candidate));
}

// Helper functions
function endCall() {
  var audios = document.getElementsByTagName("audio");
  for (var i = 0; i < audios.length; i++) {
    audios[i].pause();
  }

  pc.close();
}

function error(err) {
  endCall();
}

function startCall(){
        navigator.getUserMedia({audio: true}, function(stream) {
        pc.onaddstream({stream: stream});
       pc.addStream(stream);

        pc.createOffer(function(offer) {
                pc.setLocalDescription(new RTCSessionDescription(offer),function() {
                ws.send(JSON.stringify(offer));
                }, error, offerOptions);
        }, error);
        });
}

ws.onmessage = function(message){
        var m = JSON.parse(message.data);
        console.log(m);

        if (m.type){
                if (m.type == "offer"){
                        navigator.getUserMedia({audio: true}, function(stream) {
                        pc.onaddstream({stream: stream});
                        pc.addStream(stream);

                        pc.setRemoteDescription(new RTCSessionDescription(m), function() {
                                pc.createAnswer(function(answer) {
                                pc.setLocalDescription(new RTCSessionDescription(answer), function() {
                                        ws.send(JSON.stringify(answer));
                                }, error);
                                }, error, offerOptions);
                        }, error);
                        });
                 }
                if (m.type == "answer"){
                        pc.setRemoteDescription(new RTCSessionDescription(m), function() { }, error);
                }
        }
        if (m.candidate){
                pc.addIceCandidate(new RTCIceCandidate(m));
        }
};

要发起呼叫，您应该只调用名称正确的startCall() - 函数。

两个之间的websocket-communication（ip被删除）：

Client connecting: tcp:ip1:62322
Client connecting: tcp:ip2:50075
Text message received: {"type":"offer","sdp":"v=0\r\no=mozilla...THIS_IS_SDPARTA-45.0.2 8467526262723029465 0 IN IP4 0.0.0.0\r\ns=-\r\nt=0 0\r\na=fingerprint:sha-256 8E:FC:F3:42:12:68:95:13:98:CC:B0:8D:41:F6:4E:39:19:60:70:5A:4B:4A:9D:93:4C:A0:53:CF:58:AB:3F:A1\r\na=ice-options:trickle\r\na=msid-semantic:WMS *\r\nm=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8\r\nc=IN IP4 0.0.0.0\r\na=sendrecv\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=ice-pwd:0a7123357b97345c6f9a9474aabf0c27\r\na=ice-ufrag:5d02cec2\r\na=mid:sdparta_0\r\na=msid:{512da0cd-689a-4981-9d88-9e857ba62803} {f7e81293-42b4-44e1-9d5b-80afe2857cf8}\r\na=rtcp-mux\r\na=rtpmap:109 opus/48000/2\r\na=rtpmap:9 G722/8000/1\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=setup:actpass\r\na=ssrc:2285633480 cname:{90d0392b-c8f1-4be9-af57-8c34d08567cd}\r\n"}
Text message received: {"candidate":"candidate:0 1 UDP 2122187007 localip2 51858 typ host","sdpMid":"sdparta_0","sdpMLineIndex":0}
Text message received: {"candidate":"candidate:7 1 UDP 2122252543 ipv6-2 51859 typ host","sdpMid":"sdparta_0","sdpMLineIndex":0}
Text message received: {"candidate":"candidate:0 2 UDP 2122187006 ip2 51860 typ host","sdpMid":"sdparta_0","sdpMLineIndex":0}
Text message received: {"candidate":"candidate:7 2 UDP 2122252542 ipv6-2 52724 typ host","sdpMid":"sdparta_0","sdpMLineIndex":0}
Text message received: {"candidate":"candidate:2 1 UDP 1685987327 ip2 51858 typ srflx raddr localip2 rport 51858","sdpMid":"sdparta_0","sdpMLineIndex":0}
Text message received: {"candidate":"candidate:2 2 UDP 1685987326 ip2 51860 typ srflx raddr localip2 rport 51860","sdpMid":"sdparta_0","sdpMLineIndex":0}
Text message received: {"type":"answer","sdp":"v=0\r\no=mozilla...THIS_IS_SDPARTA-45.0.2 1868980908691513816 0 IN IP4 0.0.0.0\r\ns=-\r\nt=0 0\r\na=fingerprint:sha-256 04:2A:EB:AD:90:95:A3:A8:B8:3A:76:FE:3A:E7:DA:1F:D6:77:30:8A:87:BB:B9:3A:30:B4:9B:3D:E5:8F:58:04\r\na=ice-options:trickle\r\na=msid-semantic:WMS *\r\nm=audio 9 UDP/TLS/RTP/SAVPF 109\r\nc=IN IP4 0.0.0.0\r\na=sendrecv\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=ice-pwd:b5665ac44d79e66c392408e08bdc32ee\r\na=ice-ufrag:1a8f59a4\r\na=mid:sdparta_0\r\na=msid:{358aa02b-b035-41a8-acf6-2b787a192c60} {28e8765a-4a56-42e3-8f45-897ca7c99c05}\r\na=rtcp-mux\r\na=rtpmap:109 opus/48000/2\r\na=setup:active\r\na=ssrc:3277858225 cname:{8cee2fed-1a9e-46b9-90d7-3cad80050f3b}\r\n"}
Text message received: {"candidate":"candidate:0 1 UDP 2122252543 localip1 59706 typ host","sdpMid":"sdparta_0","sdpMLineIndex":0}
Text message received: {"candidate":"candidate:1 1 UDP 1686052863 ip1 59706 typ srflx raddr localip1 rport 59706","sdpMid":"sdparta_0","sdpMLineIndex":0}

Answer 1

更改此行：

function error(err) {
  endCall();
}

为：

function error(err) {
  console.error(err);
  endCall();
}

一般而言，您所犯的错误并不是记录InvalidStateError: Cannot add ICE candidate in state stable之类的错误。变化：

setLocalDescription

为：

offer, candidate, candidate, candidate

浏览器正试图帮助您，并经常告诉您出了什么问题。通过这种方式，下次有些东西不起作用时你就不用问了。

<强>更新

setRemoteDescription表示被叫方在收到要约之前正在接收候选人。这是WebRTC的时间敏感部分。一旦在呼叫者的呼叫中呼叫getUserMedia，冰候选者就会开始流动。例如电线看起来像这样：

setRemoteDescription

因此，在接收端，您应该立即致电getUserMedia，否则该对等连接不准备接收候选人。您的代码正在等待{{1}}，这是问题所在。

更改您的代码，以便在{{1}}之前致电{{1}}，它应该会更好。

Answer 2

您还没有配置任何TURN服务器，因此您看到候选列表中没有Relay候选者。很可能两个参与者处于使用WebRTC无法实现点对点的情况，这需要中继连接。因此配置TURN服务器可能会解决您的问题。要验证是否是这种情况，您可以尝试在两个参与者都位于相同的子网或网络后设置呼叫，并且如果可行，那么您的代码就可以了，配置TURN服务器将解决您的问题。

WebRTC RTCPeerConnection未建立

2 个答案: