WebRTC RTCPeerConnection not established

WebRTC RTCPeerConnection not established

本文关键字：established not RTCPeerConnection WebRTC      更新时间：2023-09-26

我的简单WebRTC javascript代码无法按预期工作。事实上，音频呼叫并没有建立起来（请注意，我对WebRTC了解最少，我是通过查看互联网上的示例创建的）。页面应启动两名参与者之间的音频通话。作为参与者之间的信号服务器，我使用了一个websocket服务器。此服务器仅在参与者之间中继消息。在呼叫发起过程中，通过websocket真正发送消息（一个报价、几个候选人、一个回答和其他候选人）
尽管如此，Firefox还是告诉我"ICE失败了，请参阅：webrtc了解更多详细信息"。两个参与者都在普通路由器后面
我会尽快添加我的websocket服务器的日志和about的示例：webrtc（当然是缩写）。

为什么这个代码不起作用？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 == "answerswer"){
                        pc.setRemoteDescription(new RTCSessionDescription(m), function() { }, error);
                }
        }
        if (m.candidate){
                pc.addIceCandidate(new RTCIceCandidate(m));
        }
};

要启动调用，您应该只调用名称正确的startCall()-函数。

两个给定之间的websocket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":"answerswer","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}

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，您做错的不是像err那样记录错误。更改：

function error(err) {
  endCall();
}

至：

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

浏览器正试图帮助你，并经常告诉你哪里出了问题。这样你下次就不用问SO了。

更新：

InvalidStateError: Cannot add ICE candidate in state stable意味着被叫在接收到要约之前正在接收候选者。这是WebRTC的时间敏感部分。一旦在主叫端调用了setLocalDescription，则ice候选者开始流动。例如，电线看起来是这样的：

offer, candidate, candidate, candidate

因此，在接收端，您应该立即致电setRemoteDescription，否则对等连接还没有准备好接收候选者。您的代码正在getUserMedia上等待，这是问题所在。

将代码更改为在getUserMedia之前调用setRemoteDescription，这样会更好地工作。

您尚未配置任何TURN服务器，因此您可以看到，您的候选列表中没有中继候选服务器。两个参与者很可能处于使用WebRTC无法实现对等的场景中，这需要中继连接。因此，配置TURN服务器可能会解决您的问题。为了验证这种情况，当两个参与者都在同一个子网或网络后面时，你可以尝试设置一个呼叫，如果可以进行对等，那么你的代码是可以的，配置TURN服务器将解决你的问题。

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