我使用 WebRTC 连接 2 个 Chrome 浏览器。我offer在第一个客户端上创建并将其发送signalR给第二个客户端,如下所示:
function initiate_call() {
callerPeerConn = new RTCPeerConnection(peerConnCfg);
callerPeerConn.ontrack = function (event) {
console.log('caller recived new stream');
remoteVideo.srcObject = event.streams[0];
console.log(event);
}
navigator.mediaDevices.getUserMedia({ audio: true, video: true })
.then(function (stream) {
localVideo.srcObject = stream;
for (const track of stream.getTracks()) {
callerPeerConn.addTrack(track, stream);
}
return callerPeerConn.createOffer();
})
.then(
function (offer) {
var off = new RTCSessionDescription(offer);
callerPeerConn.setLocalDescription(
new RTCSessionDescription(off),
function () {
// invite to video chat
console.log('send offer');
},
function (err) {
console.log(err.message);
}
)
});
}
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当我的第二个浏览器使用offer并setLocalDescription尝试创建answer而不是将其发送给调用者时,如下所示:
function accept_send_answer(){
calleePeerConn = new RTCPeerConnection(peerConnCfg);
calleePeerConn.ontrack = function (event) {
console.log('callee accept offer and got streams');
remoteVideo.srcObject = event.streams[0];
}
calleePeerConn.setRemoteDescription(offer)
.then(function () {
return navigator.mediaDevices.getUserMedia({ audio: true, video: true });
})
.then(function (stream) {
localVideo.srcObject = stream;
for (const track of stream.getTracks()) {
calleePeerConn.addTrack(track, stream);
}
return calleePeerConn.createAnswer();
})
.then(function (answer) {
// sending answer
console.log("sending ansfer");
var remote_streams = calleePeerConn.getRemoteStreams();
var local_streams = calleePeerConn.getLocalStreams();
console.log("callee remote streams");
console.log(remote_streams);
console.log("callee local streams");
console.log(local_streams);
})
.catch(function (err) {
console.log(err.message);
});
}
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在我更改代码后,按照受人尊敬的@jib的建议,我两侧的本地和远程流都成功添加到了 RTCPeerConnection 对象中。我在控制台中成功获得以下消息:caller recived new stream以及 callee accept offer and got streams。最后一个问题是 - 为什么这段代码不起作用:
calleePeerConn.ontrack = function (event) {
console.log('callee accept offer and got streams');
remoteVideo.srcObject = event.streams[0];
}
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视频未播放。
首先,addStream和onaddstream已被弃用,并且无法在其他浏览器中工作。请改用addTrack和ontrack。
第二,时机。
peerConn.createOffer()您之前打过电话peerConn.addStream(stream),因此不会拾取曲目。
peerConn.createAnswer()和以前一样peerConn.addStream(stream)。
最后,混合回调和承诺会混淆这里的事物顺序。尝试:
const peerConn = new RTCPeerConnection(peerConnCfg);
peerConn.ontrack = function (event) {
alert('new stream added! ' + event.streams[0]);
}
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function initiate_call() {
navigator.mediaDevices.getUserMedia({audio: true, video: true})
.then(function (stream) {
localVideo.srcObject = stream;
for (const track of stream.getTracks()) {
peerConn.addTrack(track, stream);
}
return peerConn.createOffer();
})
.then(function (offer) {
// signaling and invite
return peerConn.setLocalDescription(off);
})
.catch(function (err) {
console.log(err.message);
});
}
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function accept_send_answer(offer) {
peerConn.setRemoteDescription(offer)
.then(function () {
return navigator.mediaDevices.getUserMedia({audio: true, video: true});
})
.then(function (stream) {
video.srcObject = stream;
for (const track of stream.getTracks()) {
peerConn.addTrack(track, stream);
}
return peerConn.createAnswer();
})
.then(function (answer) {
//signaling to caller and send answer
return peerConn.setLocalDescription(answer);
})
.catch(function (err) {
console.log(err.message);
});
}
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请注意,您的代码(和我的回复)仍然缺少关键部分:冰候选人交换,并且您没有显示setRemoteDescription(answer)完成协商循环的代码。
请注意,大多数示例倾向于在双方使用相同的 JS,例如这个工作小提琴使用 iframe postMessage 进行信号发送。
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