ffmpeg - 检测为 UDP 的 RTP 数据包

Question

这是我正在尝试做的事情:

WebRTC endpoint > RTP Endpoint > ffmpeg > RTMP server.

这就是我的 SDP 文件的样子。

var cm_offer = "v=0\n" +
              "o=- 3641290734 3641290734 IN IP4 127.0.0.1\n" +
              "s=nginx\n" +
              "c=IN IP4 127.0.0.1\n" +
              "t=0 0\n" +
              "m=audio 60820 RTP/AVP 0\n" +
              "a=rtpmap:0 PCMU/8000\n" +
              "a=recvonly\n" +
              "m=video 59618 RTP/AVP 101\n" +
              "a=rtpmap:101 H264/90000\n" +
              "a=recvonly\n";

发生的事情是，wireshark 可以检测到端口 59618 的传入数据包，但不是 RTP 数据包而是 UDP 数据包。我正在尝试使用以下命令使用 ffmpeg 捕获数据包:

ubuntu@ip-132-31-40-100:~$ ffmpeg -i udp://127.0.0.1:59618 -vcodec copy stream.mp4
ffmpeg version git-2017-01-22-f1214ad Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
  configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab --enable-libwavpack --enable-nvenc
  libavutil      55. 44.100 / 55. 44.100
  libavcodec     57. 75.100 / 57. 75.100
  libavformat    57. 63.100 / 57. 63.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 69.100 /  6. 69.100
  libavresample   3.  2.  0 /  3.  2.  0
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100 

我得到的只是一个闪烁的光标，并且退出(ctrl + c)后stream.mp4文件没有写入磁盘。

所以你能帮我弄清楚:

为什么wireshark 不能将数据包检测为RTP(我怀疑它与SDP 有关)

当 RTP 端点推送到不发回答案的 ffmpeg 时如何处理 SDP 答案。

这是整个代码(修改了hello world教程)

/*
     * (C) Copyright 2014-2015 Kurento (http://kurento.org/)
     *
     * Licensed under the Apache License, Version 2.0 (the "License");
     * you may not use this file except in compliance with the License.
     * You may obtain a copy of the License at
     *
     *   http://www.apache.org/licenses/LICENSE-2.0
     *
     * Unless required by applicable law or agreed to in writing, software
     * distributed under the License is distributed on an "AS IS" BASIS,
     * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
     * See the License for the specific language governing permissions and
     * limitations under the License.
     */

    function getopts(args, opts)
    {
      var result = opts.default || {};
      args.replace(
          new RegExp("([^?=&]+)(=([^&]*))?", "g"),
          function($0, $1, $2, $3) { result[$1] = decodeURI($3); });

      return result;
    };

    var args = getopts(location.search,
    {
      default:
      {
        ws_uri: 'wss://' + location.hostname + ':8433/kurento',
        ice_servers: undefined
      }
    });

    function setIceCandidateCallbacks(webRtcPeer, webRtcEp, onerror)
    {
      webRtcPeer.on('icecandidate', function(candidate) {
        console.log("Local candidate:",candidate);

        candidate = kurentoClient.getComplexType('IceCandidate')(candidate);

        webRtcEp.addIceCandidate(candidate, onerror)
      });

      webRtcEp.on('OnIceCandidate', function(event) {
        var candidate = event.candidate;

        console.log("Remote candidate:",candidate);

        webRtcPeer.addIceCandidate(candidate, onerror);
      });
    }


    function setIceCandidateCallbacks2(webRtcPeer, rtpEp, onerror)
    {
      webRtcPeer.on('icecandidate', function(candidate) {
        console.log("Localr candidate:",candidate);

        candidate = kurentoClient.getComplexType('IceCandidate')(candidate);

        rtpEp.addIceCandidate(candidate, onerror)
      });
    }


    window.addEventListener('load', function()
    {
      console = new Console();

      var webRtcPeer;
      var pipeline;
      var webRtcEpt;

      var videoInput = document.getElementById('videoInput');
      var videoOutput = document.getElementById('videoOutput');

      var startButton = document.getElementById("start");
      var stopButton = document.getElementById("stop");

      startButton.addEventListener("click", function()
      {
        showSpinner(videoInput, videoOutput);

        var options = {
          localVideo: videoInput,
          remoteVideo: videoOutput
        };


        if (args.ice_servers) {
         console.log("Use ICE servers: " + args.ice_servers);
         options.configuration = {
           iceServers : JSON.parse(args.ice_servers)
         };
        } else {
         console.log("Use freeice")
        }

        webRtcPeer = kurentoUtils.WebRtcPeer.WebRtcPeerSendrecv(options, function(error)
        {
          if(error) return onError(error)

          this.generateOffer(onOffer)
        });

        function onOffer(error, sdpOffer)
        {
          if(error) return onError(error)

          kurentoClient(args.ws_uri, function(error, client)
          {
            if(error) return onError(error);

            client.create("MediaPipeline", function(error, _pipeline)
            {
              if(error) return onError(error);

              pipeline = _pipeline;

              pipeline.create("WebRtcEndpoint", function(error, webRtc){
                if(error) return onError(error);

                webRtcEpt = webRtc;

                setIceCandidateCallbacks(webRtcPeer, webRtc, onError)

                webRtc.processOffer(sdpOffer, function(error, sdpAnswer){
                  if(error) return onError(error);

                  webRtcPeer.processAnswer(sdpAnswer, onError);
                });
                webRtc.gatherCandidates(onError);

                webRtc.connect(webRtc, function(error){
                  if(error) return onError(error);

                  console.log("Loopback established");
                });
              });



            pipeline.create("RtpEndpoint", function(error, rtp){
                if(error) return onError(error);

                //setIceCandidateCallbacks2(webRtcPeer, rtp, onError)


                var cm_offer = "v=0\n" +
                      "o=- 3641290734 3641290734 IN IP4 127.0.0.1\n" +
                      "s=nginx\n" +
                      "c=IN IP4 127.0.0.1\n" +
                      "t=0 0\n" +
                      "m=audio 60820 RTP/AVP 0\n" +
                      "a=rtpmap:0 PCMU/8000\n" +
                      "a=recvonly\n" +
                      "m=video 59618 RTP/AVP 101\n" +
                      "a=rtpmap:101 H264/90000\n" +
                      "a=recvonly\n";



                rtp.processOffer(cm_offer, function(error, cm_sdpAnswer){
                  if(error) return onError(error);

                  //webRtcPeer.processAnswer(cm_sdpAnswer, onError);
                });
                //rtp.gatherCandidates(onError);

                webRtcEpt.connect(rtp, function(error){
                  if(error) return onError(error);

                  console.log("RTP endpoint connected to webRTC");
                });
              });









            });
          });
        }
      });
      stopButton.addEventListener("click", stop);


      function stop() {
        if (webRtcPeer) {
          webRtcPeer.dispose();
          webRtcPeer = null;
        }

        if(pipeline){
          pipeline.release();
          pipeline = null;
        }

        hideSpinner(videoInput, videoOutput);
      }

      function onError(error) {
        if(error)
        {
          console.error(error);
          stop();
        }
      }
    })


    function showSpinner() {
      for (var i = 0; i < arguments.length; i++) {
        arguments[i].poster = 'img/transparent-1px.png';
        arguments[i].style.background = "center transparent url('img/spinner.gif') no-repeat";
      }
    }

    function hideSpinner() {
      for (var i = 0; i < arguments.length; i++) {
        arguments[i].src = '';
        arguments[i].poster = 'img/webrtc.png';
        arguments[i].style.background = '';
      }
    }

    /**
     * Lightbox utility (to display media pipeline image in a modal dialog)
     */
    $(document).delegate('*[data-toggle="lightbox"]', 'click', function(event) {
      event.preventDefault();
      $(this).ekkoLightbox();
    });

Answer 1

当您使用 WebRTC 时，RTP 流使用 DTLS/SRTP 加密。
因此，您无法在 Wireshark 等网络跟踪中看到它。

我不确定 ffmpeg 是否可以正确解码它们。

您可能会尝试使用 WebRTC 网关来实现您的目标。