【发布时间】:2016-11-19 07:32:25
【问题描述】:
我在 WebRTC 的报价生成过程的抽象中遇到了一个奇怪的问题。似乎传入的冰候选者永远不会到达空候选者。在使用几乎相同的代码之前,我已经成功生成了报价,但无论出于何种原因,我的抽象版本只达到了 12 个候选人,而我原来的常规版本是 20 个。这太奇怪了,看起来代码几乎相同,但抽象的代码即使在同一个浏览器上也不起作用。
原始工作代码:
var localConn = new webkitRTCPeerConnection({'iceServers':[{...}]});
var remoteConn = new webkitRTCPeerConnection({'iceServers':[{...}]});
function initMedia(localView, callback){
navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia;
var constraints = {audio:true,video:true};
navigator.getUserMedia(constraints, successStream, errorStream);
//onSuccess and Error functions
function successStream(stream){
window.stream = stream;
if(window.URL){
$('#'+localView).attr('src',window.URL.createObjectURL(stream));
} else {
$('#'+localView).attr('src',stream);
}
localConn.addStream(stream);
remoteConn.addStream(stream);
console.log('local Stream: '+ stream.id);
callback(); //-> goes on to create new offer
}
function errorStream(error){
console.log('navigator.getUserMedia error: ', error);
}
}
//function that generates offer and sends it out to a callback
function newOffer(callback){
console.log('creating new offer');
localConn.createOffer(function (sessionDescription){
localConn.setLocalDescription(sessionDescription);
}, function(error){
console.log('Error setting local description: '+error);
});
createOffer();
//gather ICE with a callback to handle/send generated offer
function createOffer(){
localConn.onicecandidate = function(iceEvent){
console.log('gathering local ice'); //ice events fired (20 total)
//upon gathering all local ice
if(iceEvent.candidate === null){
console.log('local ice gathered'); //success
var offer = {'type': localConn.localDescription.type,
'sdp': localConn.localDescription.sdp};
offer = JSON.stringify(offer);
console.log('offer created');
callback(offer);
}
}
}
}
具有抽象功能的新版本(不获取空冰候选)
//general rtc vars
var localConn = new webkitRTCPeerConnection({'iceServers':[{'url':'stun:stun.1.google.com:19302'}]});
var remoteConn = new webkitRTCPeerConnection({'iceServers':[{'url':'stun:stun.1.google.com:19302'}]});
//var mediaStream;
var channel;
//creates a stream from webcam
//@params function streamHandle(stream)
function initStream(streamHandle){
navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia;
var constraints = {audio:true,video:true};
navigator.getUserMedia(constraints, successStream, errorStream);
//onSuccess and Error functions
function successStream(stream){
window.stream = stream;
console.log('TreRTC: Local Stream-'+ stream.id);
//mediaStream = stream;
localConn.addStream(stream); //not sure if these need to be added before
remoteConn.addStream(stream); //or if they can be added in the creatOffer(stream) function
streamHandle(stream); //gets inserted into createOffer function
}
function errorStream(error){
console.log('navigator.getUserMedia error: ', error);
}
}
//creates an offer to be sent
//@params Stream stream (from getusermedia)
//@return string offer
function createOffer(stream){
console.log('TreRTC: Creating Offer');
//localConn.addStream(stream); //tried both ways from top and from internal
//remoteConn.addStream(stream);
localConn.createOffer(function (sessionDescription){
localConn.setLocalDescription(sessionDescription);
}, function(error){
console.log('Error setting local description: '+error);
});
localConn.onicecandidate = function(iceEvt){
console.log('TreRTC: ICE in'); //ice events firing (12 total)
if(iceEvt.candidate === null){
console.log('TreRTC: ICE gathered'); //never reaches to this point...
var offer = {'type': localConn.localDescription.type,
'sdp': localConn.localDescription.sdp};
offer = JSON.stringify(offer);
console.log('TreRTC: Offer initialized');
return offer; //returns offer as a string
} //could also specify a callback
}
}
【问题讨论】:
-
请问你这个代码是基于什么例子?它使用了非常古老的结构,例如旧版回调 API,a broken createObjectURL pattern,并且仅适用于 Chrome 浏览器启动,因为 webkit 前缀(我正在尝试追踪旧示例以更新它们)。
-
stun:stun.1.google.com:19305 不是 stun 服务器。您可能想使用 stun:stun.l.google.com:19305 进行测试
标签: javascript sockets stream webrtc getusermedia