关于 node.js:webrtc onaddstream 未在第一个对等方上调用

webrtc onaddstream not being called on first peer

我创建了下面的脚本,它是混合应用程序的一部分,有时它运行正常,我可以接收/发送音频/视频呼叫,但有时甚至没有从发送方调用 onaddstreamontrack,但是 spd 数据包是通过套接字发送的,我都尝试了 (onaddstream or ontrack) 但没有成功:

在这里发送来自 pc 的报价:

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  sendOffer() {
    let that = this;
    that.call_status = 'connecting';

    let call_type;
    if (that.call_type == 'audio')
      call_type = { video: false, audio: true };
    else
      call_type = { video: true, audio: true };

    that.pc = new RTCPeerConnection(that.peerConnectionConfig);
    that.haveGum = navigator.mediaDevices.getUserMedia(call_type)
      .then(stream => {
        that.pc.addStream(that.from_video.nativeElement.srcObject = stream);
        that.from_video.nativeElement.style.display = 'block';
      }).then(() => that.pc.createOffer())
      .then(d => that.pc.setLocalDescription(d))
      .catch(log => { alert(log) });

    that.pc.oniceconnectionstatechange = function (e) {

      that.call_status = that.pc.iceConnectionState;
      if (that.pc.iceConnectionState == 'disconnected') {
        console.log('Disconnected');
      }
    }

    that.pc.onaddstream = e => {
      that.to_video.nativeElement.srcObject = e.stream;
    };


    that.pc.onicecandidate = e => {

      if (e.candidate) {
        return;
      }
        that.offerSent = true;
        that.socket.emit('sdp-offer', {
          from: that.user,
          sdp: that.pc.localDescription.sdp,
          call_type: call_type
        });
    };

    that.socket.on('sdp-offer-reply', (sdp: any) => {
      that.pc.setRemoteDescription(new RTCSessionDescription(({ type:"answer", sdp: sdp.sdp }))).catch(log => console.log(log));
    });

    that.socket.on('call-closed', (sdp: any) => {
      that.closeConnection();
    });
  }

在其他设备上 pc2 接受答案时:

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  answerCall() {
    let that = this;

    let call_type;
    if (this.call_type == 'audio')
      call_type = { video: false, audio: true };
    else
      call_type = { video: true, audio: true };

    that.pc2 = new RTCPeerConnection(this.peerConnectionConfig);
    that.haveGum = navigator.mediaDevices.getUserMedia(call_type)
      .then(stream => {
        that.pc2.addStream(this.from_video.nativeElement.srcObject = stream);
      });

    that.pc2.oniceconnectionstatechange = function (e) {
        console.log(that.pc2.iceConnectionState);
    }
    that.pc2.onaddstream = e => {
      that.to_video.nativeElement.srcObject = e.stream;
      that.to_video.nativeElement.style.display = 'block';
    };

    if (that.pc2.signalingState !="stable") {
      that.call_status = that.pc2.signalingState;
      alert("not stable");
      return;
    }

    that.pc2.setRemoteDescription(new RTCSessionDescription(({ type:"offer", sdp: this.sdp.sdp })))
      .then(() => that.pc2.createAnswer())
      .then(d => {
        that.sendSdpAnswer = d; that.pc2.setLocalDescription(d);
        this.call_connected = true;
      })
      .catch(log => console.log(log));
    that.pc2.onicecandidate = e => {
      if (e.candidate) {
        console.log("not e.candidate");
        return;
      }
      that.socket.emit('offeraccepted', {
        from: that.user,
        sdp: that.sendSdpAnswer.sdp
      });
    };


    that.socket.on('call-closed', (sdp: any) => {
      that.closeConnection();
      that.call_status ="Hung Up";
    });
  }

这是我调用的最终函数,用于在通话结束时关闭双方的对等连接:

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  closeConnection() {
    if (typeof this.pc !=="undefined" && this.pc.signalingState !="closed") {
      this.pc.close();
    }
    if (typeof this.pc2 !=="undefined" && this.pc2.signalingState !="closed") {
      this.pc2.close();
    }
  }

我正在使用 webrtc latest-adapter.js 和 socket.io 作为信号服务器。首先,我在 pc 上发出事件 以发送 sdp 数据包,在 pc2 上我从节点服务器接收 sdp-offer-incoming,然后 pc2 发出 offeraccepted 并在 pc1 上附加带有事件的 sdp 数据接收 sdp 数据包,它应该在两台电脑上显示视频/音频,但有时发送方没有收到流,但接收方总是有两个视频。


创建报价时我必须通过约束:

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      this.pc.createOffer({
        offerToReceiveAudio: 1,
        offerToReceiveVideo: 1
      })