webrtc onaddstream 未被第一个对等端调用

Question

我在下面的脚本这是一个混合应用程序的一部分创建的，有时它正常工作，并且可以接收/发送音频/视频通话，但有时onaddstream还是ontrack 不甚至从发送方调用，但SPD数据包正在通过socket发送的，我都试过了，(onaddstream or ontrack)但都没有成功：

从这里发送报价pc：

  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();
    });
  }

pc2当接受答案时，在其他设备上：

  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";
    });
  }

这是我调用的最后一个函数，用于在呼叫结束时关闭双方的对等连接：

  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();
    }
  }

我使用 webrtc latest-adapter.js 和 socket.io 作为信号服务器。首先，我EMIT事件sdp-offer上pc发送SDP包和pc2我收到sdp-offer-incoming的节点服务器，比pc2EMITofferaccepted与事件附加SDP数据，在pc1我收到的SDP数据包，并显示所行接收视频/两个PC的音频，因为它应该，但有时发件人流，但接收器总是有两个视频。

Answer 1

我在创建报价时必须通过约束：

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