[英]a few questions about icecandidate and sdp of webrtc
Suppose there has 3 users A , B , C . 假设有3个用户A,B,C。 And i am use Ajax rather than websockets to send and get messages. 我使用Ajax而不是websockets发送和获取消息。
Question blow: 问题打击:
1, In this case I just need that (B and C) can see A ,and don't need that A can see (B and C). 1,在这种情况下,我只需要(B和C)可以看到A,而不需要A可以看到(B和C)。 I have added A's sdp(offer) to B . 我已经将A的sdp(offer)添加到B中。 So is it necessary for A to add B's sdp(answer)? 那么,A有必要添加B的sdp(answer)吗?
2, I have found there has so many(about 17) icecandidate generated when A createOffer(). 2,我发现A createOffer()生成了很多(大约17个)icecandidate。 and I don't know how to handle these icecandidate . 而且我不知道如何处理这些候选人。 should B (and C) add these all icecandidate one by one for 17 times ? B(和C)是否应一一添加17种所有的冰候选物?
3, As I said 3 users. 3,正如我所说的3位用户。 This is a one-to-many application. 这是一对多的应用程序。 How to implement it? 如何执行呢? my main question is how to deal the sdp and icecandidate of each one.Does A need to add others' sdp and icecandidate . 我的主要问题是如何处理每个人的sdp和icecandidate。是否需要添加其他人的sdp和icecandidate。
here is the icecandidates blow: 这是候选人的打击:
{"candidate":"candidate:1754064501 1 udp 2122255103 2001::5ef5:79fd:8c8:c74e:f166:22f 52522 typ host generation 0","sdpMid":"audio","sdpMLineIndex":0}
{"candidate":"candidate:1026183099 1 udp 2122194687 192.168.2.100 52523 typ host generation 0","sdpMid":"audio","sdpMLineIndex":0}
{"candidate":"candidate:1754064501 2 udp 2122255102 2001::5ef5:79fd:8c8:c74e:f166:22f 52524 typ host generation 0","sdpMid":"audio","sdpMLineIndex":0}
{"candidate":"candidate:1026183099 2 udp 2122194686 192.168.2.100 52525 typ host generation 0","sdpMid":"audio","sdpMLineIndex":0}
{"candidate":"candidate:1754064501 1 udp 2122255103 2001::5ef5:79fd:8c8:c74e:f166:22f 52526 typ host generation 0","sdpMid":"video","sdpMLineIndex":1}
{"candidate":"candidate:1026183099 1 udp 2122194687 192.168.2.100 52527 typ host generation 0","sdpMid":"video","sdpMLineIndex":1}
{"candidate":"candidate:1754064501 2 udp 2122255102 2001::5ef5:79fd:8c8:c74e:f166:22f 52528 typ host generation 0","sdpMid":"video","sdpMLineIndex":1}
{"candidate":"candidate:1026183099 2 udp 2122194686 192.168.2.100 52529 typ host generation 0","sdpMid":"video","sdpMLineIndex":1}
{"candidate":"candidate:638524037 1 tcp 1518275327 2001::5ef5:79fd:8c8:c74e:f166:22f 0 typ host tcptype active generation 0","sdpMid":"audio","sdpMLineIndex":0}
{"candidate":"candidate:1940501323 1 tcp 1518214911 192.168.2.100 0 typ host tcptype active generation 0","sdpMid":"audio","sdpMLineIndex":0}
{"candidate":"candidate:638524037 2 tcp 1518275326 2001::5ef5:79fd:8c8:c74e:f166:22f 0 typ host tcptype active generation 0","sdpMid":"audio","sdpMLineIndex":0}
{"candidate":"candidate:1940501323 2 tcp 1518214910 192.168.2.100 0 typ host tcptype active generation 0","sdpMid":"audio","sdpMLineIndex":0}
{"candidate":"candidate:638524037 1 tcp 1518275327 2001::5ef5:79fd:8c8:c74e:f166:22f 0 typ host tcptype active generation 0","sdpMid":"video","sdpMLineIndex":1}
{"candidate":"candidate:1940501323 1 tcp 1518214911 192.168.2.100 0 typ host tcptype active generation 0","sdpMid":"video","sdpMLineIndex":1}
{"candidate":"candidate:638524037 2 tcp 1518275326 2001::5ef5:79fd:8c8:c74e:f166:22f 0 typ host tcptype active generation 0","sdpMid":"video","sdpMLineIndex":1}
{"candidate":"candidate:1940501323 2 tcp 1518214910 192.168.2.100 0 typ host tcptype active generation 0","sdpMid":"video","sdpMLineIndex":1}
null
and here is my application's code: you will find that i use a temp button which call "Get Answer" to let A to get sdp and icecandidate (has Commented out) of B or C 这是我的应用程序的代码:您会发现我使用一个临时按钮,调用“ Get Answer”,让A获取B或C的sdp和icecandidate(已注释掉)
var getUserMedia = (navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia || navigator.msGetUserMedia);
//use Google's stun server
var iceServer = {
"iceServers": [{
"url": "stun:stun.l.google.com:19302"
}]
};
var PeerConnection = (window.PeerConnection ||
window.webkitPeerConnection00 ||
window.webkitRTCPeerConnection ||
window.mozRTCPeerConnection);
var pc = new PeerConnection(iceServer);
var iceTimes = 1;
function getAnswer(cid)
{
teacherLoop(cid);
/*for(i=1;i<4;i++)
{
$.ajax({
url:'/get_webrtc_info.php',
type:'POST',
async:false,
data:{role:'get_answer_ice',times:i,"cid": cid},
success:function(msg)
{
var msg = JSON.parse(msg);
pc.addIceCandidate(new RTCIceCandidate(msg));
}
});
}*/
}
//start live
function startLive(cid)
{
pc.onicecandidate = function(event){
$.ajax({
url:"/webrtc.php",
type:'POST',
async:false,
data:{
"event": "__ice_candidate",
"times": iceTimes,
"cid": cid,
"data": {
"candidate": JSON.stringify(event.candidate)
}
},
});
iceTimes++;
};
$(".course-item-play-block").slideDown();
getUserMedia.call(navigator, {
"audio": true,
"video": true
}, function(stream){
var myselfVideoElement = document.getElementById('video');
myselfVideoElement.src = URL.createObjectURL(stream);
pc.addStream(stream);
pc.createOffer(function(desc){
pc.setLocalDescription(desc);
$.ajax({
url:'/webrtc.php',
type:'POST',
async:false,
data:{
"event": "__offer",
"cid": cid,
"data": {
"sdp": JSON.stringify(desc)
}
},
success:function()
{
//var timer = setInterval(teacherLoop,4000);
}
});
});
}, function(error){
//
});
}
function joinLive(cid)
{
$(".course-item-play-block").slideDown();
pc.onaddstream = function(event)
{
var remote_video = document.getElementById('video');
remote_video.src = URL.createObjectURL(event.stream);
}
pc.onicecandidate = function(event){
$.ajax({
url:"/webrtc.php",
type:'POST',
async:false,
data:{
"event": "__ice_candidate_answer",
"times": answerIceTimes,
"cid": cid,
"data": {
"candidate": JSON.stringify(event.candidate)
}
},
});
answerIceTimes++;
};
studentLoop(cid)
getIce(cid)
}
function getIce(cid)
{
for(i=1;i<17;i++)
{
$.ajax({
url:'/get_webrtc_info.php',
type:'POST',
async:false,
data:{role:'get_ice',times:i,"cid": cid},
success:function(msg)
{
var msg = JSON.parse(msg);
pc.addIceCandidate(new RTCIceCandidate(msg));
}
});
}
}
function teacherLoop(cid)
{
$.ajax({
url:'/get_webrtc_info.php',
type:'POST',
data:{role:'teacher',"cid": cid},
success:function(msg)
{
var msg = JSON.parse(msg);
pc.setRemoteDescription(new RTCSessionDescription(msg));
}
});
}
function studentLoop(cid)
{
$.ajax({
url:'/get_webrtc_info.php',
type:'POST',
async:false,
data:{role:'student',"cid": cid},
success:function(msg)
{
var msg = JSON.parse(msg);
pc.setRemoteDescription(new RTCSessionDescription(msg));
pc.createAnswer(function(answer) {
pc.setLocalDescription(new RTCSessionDescription(answer), function() {
$.ajax({
url:'/webrtc.php',
type:'POST',
async:false,
data:{
"event": "__answer",
"cid": cid,
"data": {
"sdp": JSON.stringify(answer)
}
},
success:function()
{
//var timer = setInterval(student,4000);
//studentLoop()
}
});
});
});
});
}
<div class="block course-item-play-block">
<h3>Living:</h3>
<video id="video" autoplay></video>
</div>
<?php if ($isTeacher) {?>
<div class="btn edit-course-btn common-btn common-btn-6x18 common-btn-blue" onclick="startLive(<?=$Data['cid']?>)">Start Live</div>
<div class="btn edit-course-btn common-btn common-btn-6x18 common-btn-blue" onclick="getAnswer(<?=$Data['cid']?>)">Get Answer</div>
<?php }else{?>
<div class="btn edit-course-btn common-btn common-btn-6x18 common-btn-blue" onclick="joinLive(<?=$Data['cid']?>)">Join Live</div>
<?php }?>
I have not worked on php, but as far as WebRTC is concerned: 我没有在php上工作,但是就WebRTC而言:
sdp-answer
, in fact, it doesn't matter who calls and who answers, the provider needs to add his MediaStream before making sdp offer/answer. 即使B只是消费者,而A是提供者,B仍然需要提供sdp-answer
,实际上,无论呼叫者和应答者都是什么,提供者都需要在提供sdp提供/答案之前添加其MediaStream。 PeerConnection
object on both sides. 您需要将它们传递到两侧的相应PeerConnection
对象。 without them exchanged, there is no way to start a call. 没有它们的交换,就无法发起呼叫。 PeerConnection
, You can also check out Muaz Khan's one-to-many boradcasting . 尽管是一对多的,但对于每个新消费者(B,C,...),提供者(A)需要创建一个单独的PeerConnection
,您也可以查看Muaz Khan的一对多boradcasting 。 If you want to protect the provider's bandwidth, you need to look into MCU, here is an one-to-many broadcast example from Kurento. 如果要保护提供商的带宽,则需要研究MCU, 这是Kurento的一对多广播示例。
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