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javascript - WebRTC RTCPeerConnection 未建立

转载 作者:搜寻专家 更新时间:2023-10-31 08:09:47 25 4
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我的简单 WebRTC javascript 代码没有按预期工作。事实上,音频通话并没有建立(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)。该页面应启动两个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。该服务器仅在参与者之间中继消息。在调用启动期间,消息实际上是通过 websocket 发送的(一个报价,几个候选人,一个答案和其他候选人)。
尽管如此,Firefox 还是给我“ICE 失败,有关详细信息,请参阅 about:webrtc”。两个参与者都在普通路由器后面。
我会尽快添加我的 websocket-server 的日志和一个 about:webrtc 的例子(当然缩短了)。

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));
}
};

要发起调用,您应该只调用正确命名的 startCall() 函数。

两者之间的 websocket 通信(删除了 ip):

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}

最佳答案

改变这一行:

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

到:

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

一般来说,您做错的不是记录err 之类的错误。变化:

function error(err) {
endCall();
}

到:

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

浏览器正在努力帮助您,并且经常会告诉您出了什么问题。这样下次出现问题时您就不必再询问 SO。

更新:

InvalidStateError: Cannot add ICE candidate in state stable 表示被叫方在收到报价之前正在接收候选人。这是 WebRTC 的时间敏感部分。一旦调用方调用了 setLocalDescription,ice candidates 就会开始流动。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,否则该对等连接尚未准备好接收候选人。您的代码正在等待 getUserMedia,这是问题所在。

更改您的代码以在 getUserMedia 之前调用 setRemoteDescription,它应该会工作得更好。

关于javascript - WebRTC RTCPeerConnection 未建立,我们在Stack Overflow上找到一个类似的问题: https://stackoverflow.com/questions/36751557/

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