未建立WebRTC RTCPeerConnection

Question

我的简单WebRTC javascript代码不能像预期的那样工作。事实上，音频呼叫并没有建立起来(请注意，我对WebRTC只有最低限度的了解，我是通过查看互联网上的示例创建的)。该页面应启动两个参与者之间的音频呼叫。作为参与者之间的信令服务器，我使用了websocket服务器。此服务器仅在参与者之间中继消息。在呼叫启动期间，通过websocket真正发送消息(一个报价、几个候选人、一个答案和其他候选人)。

尽管如此，Firefox还是给出了"ICE失败，请参阅about:webrtc以获取更多详细信息“。两个参与者都位于普通路由器之后。

我将尽快添加我的websocket-server的日志和一个about:webrtc (当然是缩写)的示例。

为什么这段代码不能工作？我忽略了什么？

我的代码是(记住这只在firefox中有效)：

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，您应该只调用正确命名的startCall()-function。

websocket-两个给定之间的通信(ip已删除)：

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：

    if (m.candidate){
            pc.addIceCandidate(new RTCIceCandidate(m));
    }

至：

    if (m.candidate){
            pc.addIceCandidate(new RTCIceCandidate(m.candidate))
            .catch(e => console.error(e));
    }

一般来说，您所做的错误不是像err那样记录错误。更改：

function error(err) {
  endCall();
}

至：

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

浏览器正在尝试帮助您，并且经常会告诉您哪里出了问题。这样你就不用问了，所以下一次有些东西不能工作了。

更新：

InvalidStateError: Cannot add ICE candidate in state stable意味着被呼叫者在收到提议之前就已经收到了候选人。这是WebRTC中对时间敏感的部分。一旦在调用者端调用了setLocalDescription，ice候选对象就开始流动。例如，电线看起来像这样：

offer, candidate, candidate, candidate

因此，在接收端，您应该立即调用setRemoteDescription，否则对等连接还没有准备好接收候选项。你的代码正在等待getUserMedia，这就是问题所在。

将您的代码更改为在getUserMedia之前调用setRemoteDescription，它应该会更好地工作。

Answer 2

您还没有配置任何TURN服务器，因此您看到的候选人列表中没有中继候选人。两个参与者很可能处于无法使用WebRTC进行点对点连接场景中，这需要中继连接。因此，配置TURN服务器可能会解决您的问题。要验证这种情况，您可以尝试在两个参与者位于相同的子网或网络中时建立呼叫，其中点对点应该是可能的，如果这样可以工作，那么您的代码是Ok的，配置TURN服务器将解决您的问题。