【owt】WebrtcNode, subscribe-sdp offer 流程(1)

sdp offer 流程

1. AmqpClient - New message received

sdp offer 的消息

2023-04-26T21:54:19.790  - DEBUG: AmqpClient - RpcServer New message received {
  method: 'onTransportSignaling',
  args: [
    'b149e44bb10d4e91bd162a8c6806ae7b',
    {
      sdp: 'v=0\r\n' +
        'o=- 7177131362423164715 2 IN IP4 127.0.0.1\r\n' +
        's=-\r\n' +
        't=0 0\r\n' +
        'a=group:BUNDLE 0 1\r\n' +
        'a=msid-semantic: WMS\r\n' +
        'm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 110 112 113 126\r\n' +
        'c=IN IP4 0.0.0.0\r\n' +
        'a=rtcp:9 IN IP4 0.0.0.0\r\n' +
        'a=ice-ufrag:h7T4\r\n' +
        'a=ice-pwd:F0L/DBkHlQzxDgFOclWm8vB7\r\n' +
        'a=ice-options:trickle\r\n' +
        'a=fingerprint:sha-256 5D:A4:FF:F1:C1:1B:51:19:CB:26:53:B6:49:2E:97:5F:F1:A4:B7:C7:41:68:AF:19:8D:FA:B1:D6:9D:02:60:4C\r\n' +
        'a=setup:actpass\r\n' +
        'a=mid:0\r\n' +
        'a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\n' +
        'a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\n' +
        'a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\n' +
        'a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\n' +
        'a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\n' +
        'a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\n' +
        'a=recvonly\r\n' +
        'a=rtcp-mux\r\n' +
        'a=rtpmap:111 opus/48000/2\r\n' +
        'a=rtcp-fb:111 transport-cc\r\n' +
        'a=fmtp:111 minptime=10;useinbandfec=1\r\n' +
        'a=rtpmap:103 ISAC/16000\r\n' +
        'a=rtpmap:104 ISAC/32000\r\n' +
        'a=rtpmap:9 G722/8000\r\n' +
        'a=rtpmap:102 ILBC/8000\r\n' +
        'a=rtpmap:0 PCMU/8000\r\n' +
        'a=rtpmap:8 PCMA/8000\r\n' +
        'a=rtpmap:106 CN/32000\r\n' +
        'a=rtpmap:105 CN/16000\r\n' +
        'a=rtpmap:13 CN/8000\r\n' +
        'a=rtpmap:110 telephone-event/48000\r\n' +
        'a=rtpmap:112 telephone-event/32000\r\n' +
        'a=rtpmap:113 telephone-event/16000\r\n' +
        'a=rtpmap:126 telephone-event/8000\r\n' +
        'm=video 9 UDP/TLS/RTP/SAVPF 123 114 115 116 119\r\n' +
        'c=IN IP4 0.0.0.0\r\n' +
        'a=rtcp:9 IN IP4 0.0.0.0\r\n' +
        'a=ice-ufrag:h7T4\r\n' +
        'a=ice-pwd:F0L/DBkHlQzxDgFOclWm8vB7\r\n' +
        'a=ice-options:trickle\r\n' +
        'a=fingerprint:sha-256 5D:A4:FF:F1:C1:1B:51:19:CB:26:53:B6:49:2E:97:5F:F1:A4:B7:C7:41:68:AF:19:8D:FA:B1:D6:9D:02:60:4C\r\n' +
        'a=setup:actpass\r\n' +
        'a=mid:1\r\n' +
        'a=extmap:14 urn:ietf:params:rtp-hdrext:toffset\r\n' +
        'a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\n' +
        'a=extmap:13 urn:3gpp:video-orientation\r\n' +
        'a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\n' +
        'a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\n' +
        'a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\n' +
        'a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\n' +
        'a=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07\r\n' +
        'a=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space\r\n' +
        'a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\n' +
        'a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\n' +
        'a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\n' +
        'a=recvonly\r\n' +
        'a=rtcp-mux\r\n' +
        'a=rtcp-rsize\r\n' +
        'a=rtpmap:123 AV1/90000\r\n' +
        'a=rtcp-fb:123 goog-remb\r\n' +
        'a=rtcp-fb:123 transport-cc\r\n' +
        'a=rtcp-fb:123 ccm fir\r\n' +
        'a=rtcp-fb:123 nack\r\n' +
        'a=rtcp-fb:123 nack pli\r\n' +
        'a=rtpmap:114 red/90000\r\n' +
        'a=rtpmap:115 rtx/90000\r\n' +
        'a=fmtp:115 apt=114\r\n' +
        'a=rtpmap:116 ulpfec/90000\r\n' +
        'a=rtpmap:119 rtx/90000\r\n' +
        'a=fmtp:119 apt=123\r\n',
      type: 'offer'
    }
  ],
  corrID: 39,
  replyTo: 'amq.gen-WtoELIbC4gJ1GfdYgkvSFA'
}

2. WebrtcNode - onTransportSignaling

dist/webrtc_agent/webrtc/index.js

// connectionId 应该就是operationId
that.onTransportSignaling = function (connectionId, msg, callback) {
        log.debug('onTransportSignaling, connection id:', connectionId, 'msg:', msg);
        // 通过 operationId 从mappingTransports 获取 transportId，
        // 接着通过transportId从peerConnections 获取WrtcConnection
        // 参考小结 2.1
        var conn = getWebRTCConnection(connectionId);
        // WrtcConnection conn
        if (conn) {
            // 参考小结 2.2
            conn.onSignalling(msg, connectionId);
            callback('callback', 'ok');
        } else {
          callback('callback', 'error', 'Connection does NOT exist:' + connectionId);
        }
    };

2.1 WebrtcNode - getWebRTCConnection——返回WrtcConnection

dist/webrtc_agent/webrtc/index.js

    // 返回的是WrtcConnection
    var getWebRTCConnection = function (operationId) {
        // 在WebrtcNode-createWebRTCConnection 创建
        var transportId = mappingTransports.get(operationId);
        if (peerConnections.has(transportId)) {
            return peerConnections.get(transportId);
        }
        return null;
    };

2.2 WrtcConnection.onSignalling

dist/webrtc_agent/webrtc/wrtcConnection.js

处理 offer sdp。

  that.onSignalling = function (msg, operationId) {
    var processSignalling = function () {
  // 这里type=’offer‘
      if (msg.type === 'offer') {
        log.debug('on offer:', msg.sdp);
        processOffer(msg.sdp);
      } else if (msg.type === 'candidate') {
      ...
      } else if (msg.type === 'removed-candidates') {
      ...
      }
    };
    // wrtc 是 Connection
    if (wrtc) {
      processSignalling();
    } else {
      // should not reach here
      log.error('wrtc is not ready');
    }
  };

2.3 ==========WrtcConnection.processOffer

dist/webrtc_agent/webrtc/wrtcConnection.js

这里的 remoteSdp 是null，

 const processOffer = function (sdp) {
 // 第一进来，remoteSdp 是null
    if (!remoteSdp) {
      // 1. First offer
      remoteSdp = new SdpInfo(sdp);
      // Check mid， 获取到mid 数组
      const mids = remoteSdp.mids();
      for (const mid of mids) {
         // 2. 设置finalformat
        processOfferMedia(mid);
      }
      // 3. 创建answer sdp
      localSdp = remoteSdp.answer();

      // 4. Setup transport
      let opId = null;
      for (const mid of mids) {
        if (remoteSdp.getMediaPort(mid) !== 0) {
          opId = setupTransport(mid);
        }
      }
      if (opId) {
        on_track({
          type: 'tracks-complete',
          operationId: opId
        });
      }

    } else {
     ...
    }
  };

  var remoteSdp = null;
  var localSdp = null;

2.3.1 SdpInfo.SdpInfo

dist/webrtc_agent/webrtc/sdpInfo.js

根据字符串，生成SdpInfo对象。

​

2.3.2 WrtcConnection.processOfferMedia——设置finalformat

WrtcConnection.addTrackOperation, 创建了元素operationMap

设置finalFormat

 const processOfferMedia = function (mid) {
    ...

    // Determine media format in offer
    if (remoteSdp.mediaType(mid) === 'audio') {
      const audioPreference = operationMap.get(mid).formatPreference;
      // filter audio payload
      const audioFormat = remoteSdp.filterAudioPayload(mid, audioPreference);
      operationMap.get(mid).finalFormat = audioFormat;
    } else if (remoteSdp.mediaType(mid) === 'video') {
      // 这是publish的存储的数据
      const videoPreference = operationMap.get(mid).formatPreference;
      // filter video payload    
      const videoFormat = remoteSdp.filterVideoPayload(mid, videoPreference);
      // 设置 finalFormat
      operationMap.get(mid).finalFormat = videoFormat;
    }
  };

operationMap 的元素就是WrtcConnection.addTrackOperation中添加的，这里是给finalFormat赋值

// mid => { operationId, sdpDirection, type, formatPreference, rids, enabled, finalFormat }
  var operationMap = new Map();

dist/webrtc_agent/webrtc/wrtcConnection.js，在WebrtcNode-subscirbe.md 的addTrackOperation 小节 有详细说明。

SdpInfo.filterVideoPayload

2023-05-31T16:44:11.144  - DEBUG: SdpInfo - filterVideoPayload,
mid=1,
preferenece={"optional":[{"codec":"h264"},
{"codec":"vp8"},{"codec":"vp9"},
{"codec":"av1"},{"codec":"h265"}]}

2023-05-31T16:44:11.145  - DEBUG: SdpInfo - filterVideoPayload, 
finalFmt video: {"codec":"av1"}

mediaInfo.rtp

rtp:[
                {
                    "payload":123,
                    "codec":"AV1",
                    "rate":90000
                },
                {
                    "payload":114,
                    "codec":"red",
                    "rate":90000
                },
                {
                    "payload":115,
                    "codec":"rtx",
                    "rate":90000
                },
                {
                    "payload":116,
                    "codec":"ulpfec",
                    "rate":90000
                },
                {
                    "payload":119,
                    "codec":"rtx",
                    "rate":90000
                }
            ],

filterVideoPayload(mid, preference) {
    log.debug("filterVideoPayload,mid="+ mid +",preferenece="+JSON.stringify(preference));
    // Remove unsupported profiles
    filterVP9(this.obj, preference, mid);
    const finalPrf = filterH264(this.obj, preference, mid);
    let finalFmt = null;
    let selectedPayload = -1;
    // 空
    const preferred = preference.preferred;
    const optionals = preference.optional || [];
    const relatedPayloads = new Set();
    const allowedFbTypes = [
      'ccm fir',
      'nack',
      'transport-cc',
      'goog-remb',
    ];
    const reservedCodecs = ['red', 'ulpfec'];
    const codecMap = new Map();
    const payloadOrder = new Map();

    const mediaInfo = this.media(mid);
    if (mediaInfo && mediaInfo.type == 'video') {
      let rtp, fmtp;
      let payloads;
      // Keep payload order in m line
      mediaInfo.payloads.toString().split(' ')
        .forEach((p, index) => {
          payloadOrder.set(parseInt(p), index);
        });
      if (mediaInfo.direction === 'recvonly') {
        // For subscription
        // concat(optionals.map((fmt) => fmt.codec.toLowerCase()));
        optionals.forEach((fmt) => {
          reservedCodecs.push(fmt.codec.toLowerCase());
        });
      }

      // mediaInfo.rtp 就是上面sdp里的payload
      for (let i = 0; i < mediaInfo.rtp.length; i++) {
        rtp = mediaInfo.rtp[i];
        const codec = rtp.codec.toLowerCase();
        if (reservedCodecs.includes(codec)) {
          relatedPayloads.add(rtp.payload);
        }
        codecMap.set(rtp.payload, codec);
        // 这个条件不会进入
        if (preferred && preferred.codec === codec) {
          selectedPayload = rtp.payload;
          break;
        }
        // 在optional 中找mediaInfo.rtp 的codec
        if (optionals.findIndex((fmt) => (fmt.codec === codec)) > -1) {
          if (selectedPayload < 0 ||
              payloadOrder.get(rtp.payload) < payloadOrder.get(selectedPayload)) {
            selectedPayload = rtp.payload;
          }
        }
      }
      // TODO: uncomment following code after register rtx in video receiver
      // if (!mediaInfo.simulcast) {
      //   for (i = 0; i < mediaInfo.fmtp.length; i++) {
      //     fmtp = mediaInfo.fmtp[i];
      //     if (fmtp.config.indexOf(`apt=${selectedPayload}`) > -1) {
      //       relatedPayloads.add(fmtp.payload);
      //     }
      //   }
      // }

      // Remove ulpfec if h264/h265 is selected
      const selectedCodec = codecMap.get(selectedPayload);
      if (['h264', 'h265'].includes(selectedCodec)) {
        codecMap.forEach((codec, pt) => {
          if (codec === 'ulpfec') {
            relatedPayloads.delete(pt);
          }
        });
      }
      // Remove red if vp9 SVC
      if (selectedCodec === 'vp9' && this.getLegacySimulcast(mid)) {
        codecMap.forEach((codec, pt) => {
          if (codec === 'red') {
            relatedPayloads.delete(pt);
          }
        });
      }

      relatedPayloads.add(selectedPayload);
      // Remove non-selected video payload
      mediaInfo.rtp = mediaInfo.rtp.filter(
        (rtp) => relatedPayloads.has(rtp.payload));
      if (mediaInfo.fmtp) {
        mediaInfo.fmtp = mediaInfo.fmtp.filter(
          (fmtp) => relatedPayloads.has(fmtp.payload));
      }
      if (mediaInfo.rtcpFb) {
        mediaInfo.rtcpFb = mediaInfo.rtcpFb.filter(
          (rtcp) => allowedFbTypes.includes(rtcp.type));
        mediaInfo.rtcpFb = mediaInfo.rtcpFb.filter(
          (rtcp) => relatedPayloads.has(rtcp.payload));
      }
      const payloadList = mediaInfo.payloads.toString().split(' ');
      if (selectedPayload !== -1) {
        payloadList.unshift(selectedPayload.toString());
      }
      mediaInfo.payloads = payloadList
        .filter((p) => relatedPayloads.has(parseInt(p)))
        .filter((v, index, self) => self.indexOf(v) === index)
        .join(' ');
    }

    if (selectedPayload !== -1) {
      finalFmt = { codec: codecMap.get(selectedPayload) };
      if (finalFmt.codec === 'h264' && finalPrf) {
        finalFmt.profile = finalPrf;
      }
    }
    log.debug('filterVideoPayload, finalFmt video:', JSON.stringify(finalFmt));
    return finalFmt;
  }

2.3.3 SdpInfo.answer——localSdp

通过offer到sdp，进行对应的修改，作为answer的sdp。赋值给localSdp中，在WrtcConnection.setupTransport用到。

 answer() {
    const answer = new SdpInfo(this.toString());
    answer.obj.origin = {
      username: '-',
      sessionId: '0',
      sessionVersion: 0,
      netType: 'IN',
      ipVer: 4,
      address: '127.0.0.1'
    };
    answer.obj.media.forEach(mediaInfo => {
      mediaInfo.port = 1;
      mediaInfo.rtcp = {
        port: 1,
        netType: 'IN',
        ipVer: 4,
        address: '0.0.0.0'
      };
      if (mediaInfo.setup === 'active') {
        mediaInfo.setup = 'passive';
      } else {
        mediaInfo.setup = 'active';
      }

      delete mediaInfo.iceOptions;  
      delete mediaInfo.rtcpRsize;
      if (mediaInfo.direction === 'recvonly') {
        mediaInfo.direction = 'sendonly';
      } else if (mediaInfo.direction === 'sendonly') {
        delete mediaInfo.msid;
        delete mediaInfo.ssrcGroups;
        delete mediaInfo.ssrcs;
        mediaInfo.direction = 'recvonly';
      }

      if (mediaInfo.ext && Array.isArray(mediaInfo.ext)) {
        const extMappings = [
          'urn:ietf:params:rtp-hdrext:ssrc-audio-level',
          // 'http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01',
          'urn:ietf:params:rtp-hdrext:sdes:mid',
          'urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id',
          'urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id',
          'urn:ietf:params:rtp-hdrext:toffset',
          'http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time',
          // 'urn:3gpp:video-orientation',
          // 'http://www.webrtc.org/experiments/rtp-hdrext/playout-delay',
        ];
        if (mediaInfo.type === 'video') {
          extMappings.push(
            'http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01'
          );
        }
        mediaInfo.ext = mediaInfo.ext.filter((e) => {
          return extMappings.includes(e.uri);
        });
      }

      if (mediaInfo.rids && Array.isArray(mediaInfo.rids)) {
        // Reverse rids direction
        mediaInfo.rids.forEach(r => {
          r.direction = (r.direction === 'send') ? 'recv' : 'send';
        });
      }
      if (mediaInfo.simulcast) {
        // Reverse simulcast direction
        if (mediaInfo.simulcast.dir1 === 'send') {
          mediaInfo.simulcast.dir1 = 'recv';
        } else {
          mediaInfo.simulcast.dir1 = 'send';
        }
      }
    });

    return answer;
  }

SdpInfo.getMediaPort

m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 110 112 113 126

m=video 9 UDP/TLS/RTP/SAVPF 123 114 115 116 119

9 就是port

SdpInfo.mediaType

m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 110 112 113 126

m=video 9 UDP/TLS/RTP/SAVPF 123 114 115 116 119

auido/video， 就是mediaType

operationMap——存放的是track相关的内容

dist/webrtc_agent/webrtc/wrtcConnection.js

在WebrtcNode-subscribe.md 的addTrackOperation 小节 有详细说明

  // mid => { operationId, sdpDirection, type, formatPreference, rids, enabled, finalFormat }
  var operationMap = new Map();

2.3.4 WrtcConnection.setupTransport——创建WrtcStream

dist/webrtc_agent/webrtc/wrtcConnection.js

小节 3

2.3.5 on_track——这是创建WrtcConnection 传入的callback,tracks-complete

在WebrtcNode-subscribe.md 的 3.WebrtcNode - createWebRTCConnection——返回WrtcConnection 中传入了

function onTrack(trackInfo) {
            handleTrackInfo(transportId, trackInfo, controller);
        }

调用的代码

        on_track({
          type: 'tracks-complete',
          operationId: opId
        });

详细的看3.3， 3.4 小节

3. WrtcConnection.setupTransport——创建WrtcStream

主要是创建MediaStream。

dist/webrtc_agent/webrtc/wrtcConnection.js

2023-04-26T21:54:19.800  - DEBUG: WrtcConnection - Add ssrc 1735005623 
to video in SDP for b149e44bb10d4e91bd162a8c6806ae7b

2023-04-26T21:54:19.800  - DEBUG: SdpInfo - Set SSRC: 1 [
{"id":1735005623,"attribute":"cname","value":"o/i14u9pJrxRKAsu"},
{"id":1735005623,"attribute":"msid","value":"guSQ9WN0ZG v0"},
{"id":1735005623,"attribute":"mslabel","value":"guSQ9WN0ZG"},
{"id":1735005623,"attribute":"label","value":"guSQ9WN0ZGv0"}]
2023-04-26 21:54:19,802  - DEBUG: WebRtcConnection - 
id: b149e44bb10d4e91bd162a8c6806ae7b,  message: setting remote SDP
2023-04-26 21:54:19,802  - DEBUG: WebRtcConnection - 
id: b149e44bb10d4e91bd162a8c6806ae7b,  message: processing remote SDP

const setupTransport = function (mid) {
    let rids = remoteSdp.rids(mid);
    const opSettings = operationMap.get(mid);
      // recvonly, 是out
    const direction = (opSettings.sdpDirection === 'sendonly') ? 'in' : 'out';
    const simSsrcs = remoteSdp.getLegacySimulcast(mid);
    const trackSettings = remoteSdp.getMediaSettings(mid);
    const mediaType = remoteSdp.mediaType(mid);

    trackSettings.owner = owner; // 用户id
    trackSettings.enableBWE = that.enableBWE;
    if (opSettings.finalFormat) {
      trackSettings[mediaType].format = opSettings.finalFormat;
      if (opSettings.finalFormat.codec === 'vp9' && simSsrcs) {
        // Legacy simulcast for VP9 SVC
        rids = null;
        trackSettings['video'].scalabilityMode = opSettings.scalabilityMode;
      }
    }

    if (rids) {
        ...
    } else {
        // 走这里的代码
      // No simulcast
      if (!trackMap.has(mid)) {
        // 1. Connection wrtc
        trackMap.set(mid, new WrtcStream(mid, wrtc, direction, trackSettings));
        // Set ssrc in local sdp for out direction
        // direct 是out
        if (direction === 'out') {
          // mtype = ‘video’
          const mtype = localSdp.mediaType(mid);
          // 2. 在WrtcStream 创建的时候，创建了videoFramePacketizer
          // 从WrtcStream 获取ssrc
          const ssrc = trackMap.get(mid).ssrc(mtype);
          if (ssrc) {
            log.debug(`Add ssrc ${ssrc} to ${mtype} in SDP for ${wrtcId}`);
            const opId = opSettings.operationId;
            let msid = msidMap.get(opId);
            // 3. setSsrcs
            if (msid) {
              localSdp.setSsrcs(mid, [ssrc], msid);
            } else {
              msid = localSdp.setSsrcs(mid, [ssrc]);
              msidMap.set(opId, msid);
            }
          }
        }
        // 4. Connection wrtc
        // remoteSdp 是在processOffer 创建的对象， client端发过来的sdp
        wrtc.setRemoteSdp(remoteSdp.singleMediaSdp(mid).toString(), mid);
        // 5. Notify new track， 详细见小节 3.5， 3.6
        // on_track， wrtcConnection 就是在创建的时候，传入的参数
        on_track({
          type: 'track-added',
          track: trackMap.get(mid),// WrtcStream
          operationId: opSettings.operationId,
          mid: mid
        });
      } else {
        log.warn(`Conflict trackId ${mid} for ${wrtcId}`);
      }
    }

    return opSettings.operationId;
  };

SdpInfo.rids

trackMap 属性——存放了WrtcStream

  // composedId => WrtcStream
  var trackMap = new Map();

composeId

  const composeId = function (mid, rid) {
    return mid + ':' + rid;
  };

3.1 new WrtcStream

小节 4
创建了MediaStream

见文章 WebrtcNode-subscribe-sdpOffer(2).md

>>>>>>>>>>>>不同于publish的流程>>>>>>>>>>

3.2 WrtcStream.ssrc

dist/webrtc_agent/webrtc/wrtcConnection.js

  ssrc(track) {
    if (track === 'audio' && this.audioFramePacketizer) {
      return this.audioFramePacketizer.ssrc();
    }
    if (track === 'video' && this.videoFramePacketizer) {
      return this.videoFramePacketizer.ssrc();
    }
    return null;
  }

3.2.1 addon.VideoFramePacketizer::getSsrc

source/agent/webrtc/rtcFrame/VideoFramePacketizerWrapper.cc

void VideoFramePacketizer::getSsrc(const v8::FunctionCallbackInfo<v8::Value>& args) {
  Isolate* isolate = Isolate::GetCurrent();
  HandleScope scope(isolate);

  VideoFramePacketizer* obj = ObjectWrap::Unwrap<VideoFramePacketizer>(args.Holder());
  owt_base::VideoFramePacketizer* me = obj->me;

  uint32_t ssrc = me->getSsrc();
  args.GetReturnValue().Set(Number::New(isolate, ssrc));
}

3.2.2 owt_base::VideoFramePacketizer

    uint32_t m_ssrc;
    uint32_t getSsrc() { return m_ssrc; }

bool VideoFramePacketizer::init(VideoFramePacketizer::Config& config)
{
    if (!m_videoSend) {
        // Create Send Video Stream
        ...
        m_videoSend = m_rtcAdapter->createVideoSender(sendConfig);
        m_ssrc = m_videoSend->ssrc();
        return true;
    }

    return false;
}

3.3 ？？？SdpInfo.setSsrcs

这里设置sscrs是什么作用

// ssrcs = [1735005623]
setSsrcs(mid, ssrcs, msid) {
    log.debug('setSsrcs,mid='+mid+",ssrcs="+ssrcs+",msid="+msid);
    const media = this.media(mid);
    if (!media) {
      return null;
    }
    if (!msid) {
      // Generate msid
      const alphanum = '0123456789' +
        'ABCDEFGHIJKLMNOPQRSTUVWXYZ' +
        'abcdefghijklmnopqrstuvwxyz';
      const msidLength = 10;
      msid = '';
      for (let i = 0; i < msidLength; i++) {
        msid += alphanum[Math.floor(Math.random() * alphanum.length)];
      }
    }

    const mtype = (media.type === 'audio') ? 'a' : 'v';
    // Only support one ssrc now
    const ssrc = ssrcs[0];
    media.ssrcs = [
      {id: ssrc, attribute: 'cname', value: 'o/i14u9pJrxRKAsu'},
      {id: ssrc, attribute: 'msid', value: `${msid} ${mtype}0`},
      {id: ssrc, attribute: 'mslabel', value: msid},
      {id: ssrc, attribute: 'label', value: `${msid}${mtype}0`},
    ];
    log.debug('Set SSRC:', mid, msid, JSON.stringify(media.ssrcs));
    return msid;
  }

<<<<<<<<<<<<<<<<<不同于publish的流程<<<<<<<<<<

3.4 Connection.setRemoteSdp

// streamId = mid
setRemoteSdp(sdp, streamId) {
    // webRtcConnection wrtc
    this.wrtc.setRemoteSdp(sdp, streamId || this.id);
  }

3.4.1 NAN_METHOD(WebRtcConnection::setRemoteSdp)

source/agent/webrtc/rtcConn/WebRtcConnection.cc

NAN_METHOD(WebRtcConnection::setRemoteSdp) {
  WebRtcConnection* obj = Nan::ObjectWrap::Unwrap<WebRtcConnection>(info.Holder());
  std::shared_ptr<erizo::WebRtcConnection> me = obj->me;
  if (!me) {
    return;
  }

  std::string sdp = getString(info[0]);
  std::string stream_id = getString(info[1]);

  bool r = me->setRemoteSdp(sdp, stream_id);

  info.GetReturnValue().Set(Nan::New(r));
}

3.4.2 erizo::WebRtcConnection::setRemoteSdp

source/agent/webrtc/rtcConn/erizo/src/erizo/WebRtcConnection.cpp

2023-04-26 21:54:19,802  - DEBUG: WebRtcConnection - 
id: b149e44bb10d4e91bd162a8c6806ae7b, 
 message: setting remote SDP

bool WebRtcConnection::setRemoteSdpInfo(std::shared_ptr<SdpInfo> sdp, std::string stream_id) {
  asyncTask([sdp, stream_id] (std::shared_ptr<WebRtcConnection> connection) {
    ELOG_DEBUG("%s message: setting remote SDPInfo", connection->toLog());

    if (!connection->sending_) {
      return;
    }

    connection->remote_sdp_ = sdp;
    // mid 
    connection->processRemoteSdp(stream_id);
  });
  return true;
}

??? erizo::WebRtcConnection::processRemoteSdp

source/agent/webrtc/rtcConn/erizo/src/erizo/WebRtcConnection.cpp

2023-04-26 21:54:19,802  - DEBUG: WebRtcConnection - id: b149e44bb10d4e91bd162a8c6806ae7b,  message: processing remote SDP

见后文owt-server/WebRTCEvent.md

SdpInfo.singleMediaSdp

dist-debug/webrtc_agent/webrtc/sdpInfo.js

  singleMediaSdp(mid) {
    const sdp = new SdpInfo(this.toString());
    sdp.obj.media = sdp.obj.media.filter(m => m.mid.toString() === mid);
    sdp.setBundleMids([mid]);
    return sdp;
  }

3.5 on__track——track add

dist/webrtc_agent/webrtc/wrtcConnection.js

通知到WebrtcNode（/dist/webrtc_agent/webrtc/index.js）的createWebRTCConnection 中注册的callback, 回调一个对象，

        // Notify new track
        on_track({
          type: 'track-added',
          track: trackMap.get(mid), // WrtcStream
          operationId: opSettings.operationId,
          mid: mid
        })

3.6 ======WebrtcNode.handleTrackInfo

dist/webrtc_agent/webrtc/index.js

// 这里传入onTrack， 就是on_track，就是 callback
var connection = new WrtcConnection({
            connectionId: transportId,
            threadPool: threadPool,
            ioThreadPool: ioThreadPool,
            network_interfaces: global.config.webrtc.network_interfaces,
            owner,
        }, function onTransportStatus(status) {
            notifyTransportStatus(controller, transportId, status);
        }, function onTrack(trackInfo) {
            /* trackInfo 就是 {
              type: 'track-added',
              track: trackMap.get(mid), // WrtcStream
              operationId: opSettings.operationId,
              mid: mid
            }*/
            handleTrackInfo(transportId, trackInfo, controller);
        });

// trackInfo 就是on__track 回调回来
var handleTrackInfo = function (transportId, trackInfo, controller) {
        var publicTrackId;
        var updateInfo;
        if (trackInfo.type === 'track-added') {
            // Generate public track ID
            const track = trackInfo.track; // WrtcStream
            publicTrackId = transportId + '-' + track.id;
            if (mediaTracks.has(publicTrackId)) {
                log.error('Conflict public track id:', publicTrackId, transportId, track.id);
                return;
            }
            mediaTracks.set(publicTrackId, track);
            mappingPublicId.get(transportId).set(track.id, publicTrackId);
            if (track.direction === 'in') {
                const trackSource = track.sender();
                router.addLocalSource(publicTrackId, 'webrtc', trackSource)
                .catch(e => log.warn('Unexpected error during track add:', e));
            } else {
                // 走这里
                router.addLocalDestination(publicTrackId, 'webrtc', track)
                .catch(e => log.warn('Unexpected error during track add:', e));
            }

            // Bind media-update handler
            track.on('media-update', (jsonUpdate) => {
                log.debug('notifyMediaUpdate:', publicTrackId, jsonUpdate);
                notifyMediaUpdate(controller, publicTrackId, track.direction, JSON.parse(jsonUpdate));
            });
            // Notify controller
            const mediaType = track.format('audio') ? 'audio' : 'video';
            updateInfo = {
                type: 'track-added',
                trackId: publicTrackId,
                mediaType: track.format('audio') ? 'audio' : 'video',
                mediaFormat: track.format(mediaType),
                direction: track.direction,
                operationId: trackInfo.operationId,
                mid: trackInfo.mid,
                rid: trackInfo.rid,
                active: true,
            };
            log.debug('notifyTrackUpdate', controller, publicTrackId, updateInfo);
            notifyTrackUpdate(controller, transportId, updateInfo);

        } else if (trackInfo.type === 'track-removed') {
            publicTrackId = mappingPublicId.get(transportId).get(trackInfo.trackId);
            if (!mediaTracks.has(publicTrackId)) {
                log.error('Non-exist public track id:', publicTrackId, transportId, trackInfo.trackId);
                return;
            }
            log.debug('track removed:', publicTrackId);
            router.removeConnection(publicTrackId)
            .then(ok => {
                mediaTracks.get(publicTrackId).close();
                mediaTracks.delete(publicTrackId);
                mappingPublicId.get(transportId).delete(trackInfo.trackId);
            })
            .catch(e => log.warn('Unexpected error during track remove:', e));

            // Notify controller
            updateInfo = {
                type: 'track-removed',
                trackId: publicTrackId,
            };
            notifyTrackUpdate(controller, transportId, updateInfo);

        } else if (trackInfo.type === 'tracks-complete') {
            updateInfo = {
                type: 'tracks-complete',
                operationId: trackInfo.operationId
            };
            notifyTrackUpdate(controller, transportId, updateInfo);
        }
    };

详细说明见后文 WebrtcNode-subscribe-sdpOffer(3).md的2. WebrtcNode.handleTrackInfo 。

流程图

文章来源地址https://www.toymoban.com/news/detail-487805.html

到了这里，关于【owt】WebrtcNode, subscribe-sdp offer 流程(1)的文章就介绍完了。如果您还想了解更多内容，请在右上角搜索TOY模板网以前的文章或继续浏览下面的相关文章，希望大家以后多多支持TOY模板网！