我使用WebRTC连接了2个Chrome浏览器。我在第一个上创建offer
,然后通过signalR
将其发送给第二个客户端:
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);
}
)
});
}
当我的第二个浏览器获得offer
时,他使用setLocalDescription
并尝试创建answer
,而不是像这样将其发送给呼叫者:
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);
});
}
更改代码后,遵循受人尊敬的@jib的建议,将我双方的本地和远程流成功添加到RTCPeerConnection对象。我也成功在控制台中获取了以下消息:caller recived new stream
和callee accept offer and got streams
。最后一个问题是-为什么此代码不起作用:
calleePeerConn.ontrack = function (event) {
console.log('callee accept offer and got streams');
remoteVideo.srcObject = event.streams[0];
}
视频未播放。
答案 0 :(得分:2)
首先,addStream
和onaddstream
已过时,在其他浏览器中不起作用。改用addTrack和ontrack。
第二,时间。
您在peerConn.createOffer()
之前叫peerConn.addStream(stream)
,所以不会拾取音轨。
与peerConn.createAnswer()
之前的peerConn.addStream(stream)
相同。
最后,混合使用回调和Promise会混淆这里的顺序。试试:
const peerConn = new RTCPeerConnection(peerConnCfg);
peerConn.ontrack = function (event) {
alert('new stream added! ' + event.streams[0]);
}
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);
});
}
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);
});
}
请注意,您的代码(和我的回复)仍然缺少关键内容:进行候选人交换,并且您没有显示setRemoteDescription(answer)
代码来完成协商循环。
请注意,大多数示例都倾向于在两侧使用相同的JS,例如working fiddle使用iframe postMessage进行信号传输。