node.js - webrtc onaddstream 未在第一个对等点上调用

Question

我创建了以下脚本，它是混合应用程序的一部分，有时它可以正常运行，我可以接收/发送音频/视频调用，但有时 onaddstream 或 ontrack甚至没有从发送方调用，但 spd 数据包是通过套接字发送的，我已经尝试了 (onaddstream 或 ontrack) 但没有成功:

在此从 pc 发送报价:

  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 pc2 上接受答案时:

  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:

  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 webrtclatest-adapter.js 和 socket.io 作为信令服务器。首先，我在 pc 上发出事件 sdp-offer 以发送 sdp 数据包，并在 pc2 上接收 sdp-offer-incoming 来自 Node 服务器，然后 pc2 发出 offeraccepted 并将 sdp 数据附加到事件，在 pc1 上我收到 sdp 数据包并显示视频/两台电脑上的音频都正常，但有时发送者没有收到流，但接收者总是有两个视频。

Answer 1

创建报价时我必须传递约束:

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