Question

我在linux上使用google chrome 21.x，webrtc对等连接已建立，但无法接收任何远程视频流，给予peerconnection“.onaddstream”的回调永远不会被调用，有人可以建议我在哪里需要看？

我正在粘贴我的整个代码，仍然无法接收远程视频流，也没有任何错误。

var peerConnCreated = false;
var peerConn = null;
var cameraOn = false;
var clientId = 0;
var svcName = "";
var clientIdRecvd = false;
var myname = "";
var hisname = "";
var myJsep;
var hisJsep;
var mySdp;
var hisSdp;

function login()
{
    var loginid = document.getElementById("login").value;
    var jsonText = {"clientid":clientId, "service":"rtc", "mtype": "online", "username": loginid};
    myname = loginid;
    socket.send(JSON.stringify(jsonText));
}

function iceCallback(canditate, moreToFollow)
{
    if(canditate) {
        console.log("ice canditate");
        var jsonText = {"clientid":clientId, "service":"rtc", "mtype": "canditate", "sndr": myname, "rcpt": hisname, 
            "label": canditate.label, "cand": canditate.toSdp()};
        socket.send(JSON.stringify(jsonText));
    }
}

function onSessionConnecting(message)
{
    console.log("session connecting ...");
}

function onRemoteStreamRemoved(event)
{
    console.log("remote stream removed");
    remotevid.src = "";
}

function onSessionOpened(message)
{
    console.log("session opened");
}

function onRemoteStreamAdded(event)
{
    console.log("remote stream added");
    remotevid.src = window.webkitURL.createObjectURL(event.stream);
    remotevid.style.opacity = 1;
}

function createPeerConnection()
{
    if (peerConnCreated) return;
    peerConn = new webkitPeerConnection00("STUN stun.l.google.com:19302", iceCallback); 
    peerConn.onconnecting = onSessionConnecting;
    peerConn.onopen = onSessionOpened;
    peerConn.onaddstream = onRemoteStreamAdded;
    peerConn.onremovestream = onRemoteStreamRemoved;
    console.log("peer connection created");
    peerConnCreated = true;
}

function turnOnCameraAndMic()
{
    navigator.webkitGetUserMedia({video:true, audio:true}, successCallback, errorCallback);
    function successCallback(stream) {
        sourcevid.style.opacity = 1;
        sourcevid.src = window.webkitURL.createObjectURL(stream);
        peerConn.addStream(stream);
        console.log("local stream added");
    }
    function errorCallback(error) {
        console.error('An error occurred: [CODE ' + error.code + ']');
    }
    cameraOn = true;
}

function dialUser(user)
{
    if (!peerConnCreated) createPeerConnection();
    hisname = user;
    var localOffer = peerConn.createOffer({has_audio:true, has_video:true});
    peerConn.setLocalDescription(peerConn.SDP_OFFER, localOffer);
    mySdp =  peerConn.localDescription;
    myJsep = mySdp.toSdp();
    var call = {"clientid":clientId, "service":"rtc", "mtype": "call", "sndr": myname, "rcpt": hisname, "jsepdata": myJsep};
    socket.send(JSON.stringify(call));
    console.log("sent offer");
    //console.log(myJsep);
    peerConn.startIce();
    console.log("ice started ");
}

//handle the message from the sip server
//There is a new connection from our peer so turn on the camera 
//and relay the stream to peer.
function handleRtcMessage(request)
{
    var sessionRequest = eval('(' + request + ')');
    switch(sessionRequest.mtype) 
    {
        case 'online':
            console.log("new user online");
            var newuser = sessionRequest.username;
            var li = document.createElement("li");
            var name = document.createTextNode(newuser);
            li.appendChild(name);
            li.onclick = function() { dialUser(newuser); };
            document.getElementById("Contact List").appendChild(li);
            break;

        case 'call':
            console.log("recvng call");
            alert("Incoming call ...");
            if (!peerConnCreated) createPeerConnection();
            peerConn.setRemoteDescription(peerConn.SDP_OFFER, new SessionDescription(sessionRequest.jsepdata));
            hisname = sessionRequest.sndr;
            var remoteOffer = peerConn.remoteDescription;
            //console.log("remoteOffer" + remoteOffer.toSdp());
            var localAnswer = peerConn.createAnswer(remoteOffer.toSdp(), {has_audio:true, has_video:true}); 
            peerConn.setLocalDescription(peerConn.SDP_ANSWER, localAnswer);
            var jsonText = {"clientid":clientId,"service":"rtc", "mtype": "pickup", "sndr" :myname, "rcpt": hisname, "jsepdata": localAnswer.toSdp()};
            socket.send(JSON.stringify(jsonText));
            console.log("sent answer");
            //console.log(localAnswer.toSdp());
            peerConn.startIce();
            if (!cameraOn) turnOnCameraAndMic();
            break;

        case 'pickup':
            console.log("recvd pickup");
            peerConn.setRemoteDescription(peerConn.SDP_ANSWER, new SessionDescription(sessionRequest.jsepdata));
            hisname = sessionRequest.sndr;
            if (!cameraOn) turnOnCameraAndMic();
            break;

        case 'canditate':
            console.log("recvd canditate");
            var canditate = new IceCandidate(sessionRequest.label, sessionRequest.cand);
            peerConn.processIceMessage(canditate);
            break;

        case 'bye':
            console.log("recvd bye");
            break;
    }
}

//open the websocket  to the antkorp webserver
var socket = new WebSocket('ws://bldsvrub:9981');
var sourcevid = null;
var remotevid = null;

socket.onopen = function () {
    console.log("websocket opened");
    sourcevid = document.getElementById("sourcevid");
    remotevid = document.getElementById("remotevid");
};

socket.onmessage = function (event) { 
    if (!clientIdRecvd) {
        var reqObj = eval('(' + event.data + ')');
        clientId = reqObj.clientid;
        svcName  = reqObj.service;
        clientIdRecvd = true;
    } else {
        //hookup the new handler to process session requests
        handleRtcMessage(event.data);
    }
};

socket.onclose = function (event) { socket = null; };

Answer 1

上面粘贴的代码包含一个小bug，应该在生成答案或商品之前将流添加到对等连接，即＆＃34; addStream＆＃34;应该在任何setlocalDescription或setRemoteDescription调用之前调用。

Answer 2

许多WebRTC演示：

https://github.com/muaz-khan/WebRTC-Experiment

E.g。一对一WebRTC音频/视频/屏幕调用：

https://github.com/muaz-khan/WebRTC-Experiment#one-to-one-calls

注意：

这个问题是TOOO-Old。这就是为什么我不认为我应该在这里添加一个工作片段代码片段。以上链接回答了所有问题。

但是，如果你是NEW-WebRTC用户并且你面临类似的问题，那么这里有一些提示：

在创建对等体之前，请确保两个对等体都已准备好进行握手。
就绪意味着，两个对等方都可以访问媒体流（音频和/或视频）
第一个对等体应该启动RTCPeerConnection对象，调用“addStream”并创建商品描述。
第二个对等方应该从第一个对等方接收OFFER-SDP。
第二个对等体应该在创建ANSWER-description之前启动RTCPeerConnection对象，先调用“addStream”和setRemoteDescription。
第二个同行应该创建ANSWER-SDP。
第一个对等方应该获得ANSWER-SDP并设置远程描述。
ICE候选对应与上述过程并行交换。

你可以在这里找到一些教程：

https://www.webrtc-experiment.com/#documentations

记住

此答案针对WebRTC-1.0。它没有回答WebRTC-1.1（ORTC）或更新的版本。

Answer 3

应该在收到包含至少一个流的答案时调用 onaddstream。如果您没有收到回调，请确保已调用setLocal和setRemoteDescription并成功。

Answer 4

我发现，就我而言，如果不回答其他视频，我将无法接收视频。

我通过伪造的视频流来解决：

            let w = 640;
            let h = 480;
            let canvas: any = Object.assign(document.createElement("canvas"), { w, h });
            canvas.getContext('2d').fillRect(0, 0, w, h);
            let blackStream = canvas.captureStream();
            outgoingStream.addTrack(blackStream.getVideoTracks()[0]);

我无法接收远程视频流

4 个答案:

注意：

记住