werift 0.18.1 → 0.18.2

package/lib/webrtc/src/media/receiver/nack.js.map

@@ -1 +1 @@
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debug from \"debug\";\nimport range from \"lodash/range\";\nimport Event from \"rx.mini\";\n\nimport { uint16Add } from \"../../../../common/src\";\nimport {\n  GenericNack,\n  RtcpTransportLayerFeedback,\n  RtpPacket,\n} from \"../../../../rtp/src\";\nimport { RTCRtpReceiver } from \"../rtpReceiver\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/receiver/nack.ts\");\n\nconst LOST_SIZE = 30 * 5;\n\nexport class NackHandler {\n  private newEstSeqNum = 0;\n  private _lost: { [seqNum: number]: number } = {};\n  private nackLoop: any;\n\n  readonly onPacketLost = new Event<[GenericNack]>();\n  mediaSourceSsrc?: number;\n  retryCount = 10;\n  closed = false;\n\n  constructor(private receiver: RTCRtpReceiver) {}\n\n  get lostSeqNumbers() {\n    return Object.keys(this._lost).map(Number).sort();\n  }\n\n  private getLost(seq: number) {\n    return this._lost[seq];\n  }\n\n  private setLost(seq: number, count: number) {\n    this._lost[seq] = count;\n\n    if (this.nackLoop || this.closed) {\n      return;\n    }\n    this.nackLoop = setInterval(async () => {\n      try {\n        await this.sendNack();\n        if (!Object.keys(this._lost).length) {\n          clearInterval(this.nackLoop);\n          this.nackLoop = undefined;\n        }\n      } catch (error) {\n        log(\"failed to send nack\", error);\n      }\n    }, 5);\n  }\n\n  private removeLost(sequenceNumber: number) {\n    delete this._lost[sequenceNumber];\n  }\n\n  addPacket(packet: RtpPacket) {\n    const { sequenceNumber, ssrc } = packet.header;\n    this.mediaSourceSsrc = ssrc;\n\n    if (this.newEstSeqNum === 0) {\n      this.newEstSeqNum = sequenceNumber;\n      return;\n    }\n\n    if (this.getLost(sequenceNumber)) {\n      // log(\"packetLoss resolved\", { sequenceNumber });\n      this.removeLost(sequenceNumber);\n      return;\n    }\n\n    if (sequenceNumber === uint16Add(this.newEstSeqNum, 1)) {\n      this.newEstSeqNum = sequenceNumber;\n    } else if (sequenceNumber > uint16Add(this.newEstSeqNum, 1)) {\n      // packet lost detected\n      range(uint16Add(this.newEstSeqNum, 1), sequenceNumber).forEach((seq) => {\n        this.setLost(seq, 1);\n      });\n      // this.receiver.sendRtcpPLI(this.mediaSourceSsrc);\n\n      this.newEstSeqNum = sequenceNumber;\n      this.pruneLost();\n    }\n  }\n\n  private pruneLost() {\n    if (this.lostSeqNumbers.length > LOST_SIZE) {\n      this._lost = Object.entries(this._lost)\n        .slice(-LOST_SIZE)\n        .reduce((acc, [key, v]) => {\n          acc[key] = v;\n          return acc;\n        }, {} as { [seqNum: number]: number });\n    }\n  }\n\n  close() {\n    this.closed = true;\n    clearInterval(this.nackLoop);\n    this._lost = {};\n  }\n\n  private updateRetryCount() {\n    this.lostSeqNumbers.forEach((seq) => {\n      const count = this._lost[seq]++;\n      if (count > this.retryCount) {\n        this.removeLost(seq);\n        return seq;\n      }\n    });\n  }\n\n  private sendNack = () =>\n    new Promise((r, f) => {\n      if (this.lostSeqNumbers.length > 0 && this.mediaSourceSsrc) {\n        const nack = new GenericNack({\n          senderSsrc: this.receiver.rtcpSsrc,\n          mediaSourceSsrc: this.mediaSourceSsrc,\n          lost: this.lostSeqNumbers,\n        });\n        // log(\"sendNack\", nack.toJSON());\n        const rtcp = new RtcpTransportLayerFeedback({\n          feedback: nack,\n        });\n        this.receiver.dtlsTransport.sendRtcp([rtcp]).then(r).catch(f);\n\n        this.updateRetryCount();\n        this.onPacketLost.execute(nack);\n      }\n    });\n}\n"]}
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debug from \"debug\";\nimport range from \"lodash/range\";\nimport Event from \"rx.mini\";\n\nimport { uint16Add } from \"../../../../common/src\";\nimport {\n  GenericNack,\n  RtcpTransportLayerFeedback,\n  RtpPacket,\n} from \"../../../../rtp/src\";\nimport { RTCRtpReceiver } from \"../rtpReceiver\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/receiver/nack.ts\");\n\nconst LOST_SIZE = 30 * 5;\n\nexport class NackHandler {\n  private newEstSeqNum = 0;\n  private _lost: { [seqNum: number]: number } = {};\n  private nackLoop: any;\n\n  readonly onPacketLost = new Event<[GenericNack]>();\n  mediaSourceSsrc?: number;\n  retryCount = 10;\n  closed = false;\n\n  constructor(private receiver: RTCRtpReceiver) {}\n\n  get lostSeqNumbers() {\n    return Object.keys(this._lost).map(Number).sort();\n  }\n\n  private getLost(seq: number) {\n    return this._lost[seq];\n  }\n\n  private setLost(seq: number, count: number) {\n    this._lost[seq] = count;\n\n    if (this.nackLoop || this.closed) {\n      return;\n    }\n    this.nackLoop = setInterval(async () => {\n      try {\n        await this.sendNack();\n        if (!Object.keys(this._lost).length) {\n          clearInterval(this.nackLoop);\n          this.nackLoop = undefined;\n        }\n      } catch (error) {\n        log(\"failed to send nack\", error);\n      }\n    }, 5);\n  }\n\n  private removeLost(sequenceNumber: number) {\n    delete this._lost[sequenceNumber];\n  }\n\n  addPacket(packet: RtpPacket) {\n    const { sequenceNumber, ssrc } = packet.header;\n    this.mediaSourceSsrc = ssrc;\n\n    if (this.newEstSeqNum === 0) {\n      this.newEstSeqNum = sequenceNumber;\n      return;\n    }\n\n    if (this.getLost(sequenceNumber)) {\n      // log(\"packetLoss resolved\", { sequenceNumber });\n      this.removeLost(sequenceNumber);\n      return;\n    }\n\n    if (sequenceNumber === uint16Add(this.newEstSeqNum, 1)) {\n      this.newEstSeqNum = sequenceNumber;\n    } else if (sequenceNumber > uint16Add(this.newEstSeqNum, 1)) {\n      // packet lost detected\n      range(uint16Add(this.newEstSeqNum, 1), sequenceNumber).forEach((seq) => {\n        this.setLost(seq, 1);\n      });\n      // this.receiver.sendRtcpPLI(this.mediaSourceSsrc);\n\n      this.newEstSeqNum = sequenceNumber;\n      this.pruneLost();\n    }\n  }\n\n  private pruneLost() {\n    if (this.lostSeqNumbers.length > LOST_SIZE) {\n      this._lost = Object.entries(this._lost)\n        .slice(-LOST_SIZE)\n        .reduce((acc, [key, v]) => {\n          acc[key] = v;\n          return acc;\n        }, {} as { [seqNum: number]: number });\n    }\n  }\n\n  close() {\n    this.closed = true;\n    clearInterval(this.nackLoop);\n    this._lost = {};\n  }\n\n  private updateRetryCount() {\n    this.lostSeqNumbers.forEach((seq) => {\n      const count = this._lost[seq]++;\n      if (count > this.retryCount) {\n        this.removeLost(seq);\n        return seq;\n      }\n    });\n  }\n\n  private sendNack = () =>\n    new Promise((r, f) => {\n      if (this.lostSeqNumbers.length > 0 && this.mediaSourceSsrc) {\n        const nack = new GenericNack({\n          senderSsrc: this.receiver.rtcpSsrc,\n          mediaSourceSsrc: this.mediaSourceSsrc,\n          lost: this.lostSeqNumbers,\n        });\n\n        const rtcp = new RtcpTransportLayerFeedback({\n          feedback: nack,\n        });\n        this.receiver.dtlsTransport.sendRtcp([rtcp]).then(r).catch(f);\n\n        this.updateRetryCount();\n        this.onPacketLost.execute(nack);\n      }\n    });\n}\n"]}

package/lib/webrtc/src/media/rtpReceiver.d.ts

@@ -70,4 +70,3 @@ export declare class RTCRtpReceiver {
     handleRtpByRid: (packet: RtpPacket, rid: string, extensions: Extensions) => void;
     private handleRTP;
 }
-export declare function unwrapRtx(rtx: RtpPacket, payloadType: number, ssrc: number): RtpPacket;

package/lib/webrtc/src/media/rtpReceiver.js

@@ -3,9 +3,8 @@ var __importDefault = (this && this.__importDefault) || function (mod) {
     return (mod && mod.__esModule) ? mod : { "default": mod };
 };
 Object.defineProperty(exports, "__esModule", { value: true });
-exports.unwrapRtx = exports.RTCRtpReceiver = void 0;
+exports.RTCRtpReceiver = void 0;
 const debug_1 = require("debug");
-const jspack_1 = require("jspack");
 const rx_mini_1 = __importDefault(require("rx.mini"));
 const promises_1 = require("timers/promises");
 const uuid_1 = require("uuid");
@@ -386,7 +385,7 @@ class RTCRtpReceiver {
             const rtxCodec = this.codecs[codecParams["apt"]];
             if (packet.payload.length < 2)
                 return;
-            packet = unwrapRtx(packet, rtxCodec.payloadType, originalSsrc);
+            packet = (0, src_2.unwrapRtx)(packet, rtxCodec.payloadType, originalSsrc);
             track = this.trackBySSRC[originalSsrc];
         }
         let red;
@@ -418,15 +417,4 @@ class RTCRtpReceiver {
     }
 }
 exports.RTCRtpReceiver = RTCRtpReceiver;
-function unwrapRtx(rtx, payloadType, ssrc) {
-    const packet = new src_2.RtpPacket(new src_2.RtpHeader({
-        payloadType,
-        marker: rtx.header.marker,
-        sequenceNumber: jspack_1.jspack.Unpack("!H", rtx.payload.subarray(0, 2))[0],
-        timestamp: rtx.header.timestamp,
-        ssrc,
-    }), rtx.payload.subarray(2));
-    return packet;
-}
-exports.unwrapRtx = unwrapRtx;
 //# sourceMappingURL=rtpReceiver.js.map

package/lib/webrtc/src/media/rtpReceiver.js.map

@@ -1 +1 @@
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{ debug } from \"debug\";\nimport { jspack } from \"jspack\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\n\nimport { int } from \"../../../common/src\";\nimport {\n  PictureLossIndication,\n  Red,\n  RedHandler,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  RtpHeader,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { codecParametersFromString, PeerConfig, usePLI, useTWCC } from \"..\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, timestampSeconds } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport { RTCRtpCodecParameters, RTCRtpReceiveParameters } from \"./parameters\";\nimport { NackHandler } from \"./receiver/nack\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { StreamStatistics } from \"./receiver/statistics\";\nimport { Extensions } from \"./router\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpReceiver.ts\");\n\nexport class RTCRtpReceiver {\n  private readonly codecs: { [pt: number]: RTCRtpCodecParameters } = {};\n  private get codecArray() {\n    return Object.values(this.codecs).sort(\n      (a, b) => a.payloadType - b.payloadType\n    );\n  }\n  private readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new NackHandler(this);\n  private readonly audioRedHandler = new RedHandler();\n\n  readonly type = \"receiver\";\n  readonly uuid = uuid();\n  readonly tracks: MediaStreamTrack[] = [];\n  readonly trackBySSRC: { [ssrc: string]: MediaStreamTrack } = {};\n  readonly trackByRID: { [rid: string]: MediaStreamTrack } = {};\n  /**last sender Report Timestamp\n   * compactNtp\n   */\n  readonly lastSRtimestamp: { [ssrc: number]: number } = {};\n  /**seconds */\n  readonly receiveLastSRTimestamp: { [ssrc: number]: number } = {};\n  readonly onPacketLost = this.nack.onPacketLost;\n  readonly onRtcp = new Event<[RtcpPacket]>();\n\n  dtlsTransport!: RTCDtlsTransport;\n  sdesMid?: string;\n  latestRid?: string;\n  latestRepairedRid?: string;\n\n  receiverTWCC?: ReceiverTWCC;\n  stopped = false;\n  remoteStreamId?: string;\n  remoteTrackId?: string;\n\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n  private remoteStreams: { [ssrc: number]: StreamStatistics } = {};\n\n  constructor(\n    readonly config: PeerConfig,\n    public kind: Kind,\n    public rtcpSsrc: number\n  ) {}\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.dtlsTransport = dtls;\n  }\n\n  // todo fix\n  get track() {\n    return this.tracks[0];\n  }\n\n  get nackEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find((f) => f.type === \"nack\");\n  }\n\n  get twccEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === useTWCC().type\n    );\n  }\n\n  get pliEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === usePLI().type\n    );\n  }\n\n  prepareReceive(params: RTCRtpReceiveParameters) {\n    params.codecs.forEach((c) => {\n      this.codecs[c.payloadType] = c;\n    });\n    params.encodings.forEach((e) => {\n      if (e.rtx) {\n        this.ssrcByRtx[e.rtx.ssrc] = e.ssrc;\n      }\n    });\n  }\n\n  /**\n   * setup TWCC if supported\n   */\n  setupTWCC(mediaSourceSsrc: number) {\n    if (this.twccEnabled && !this.receiverTWCC) {\n      this.receiverTWCC = new ReceiverTWCC(\n        this.dtlsTransport,\n        this.rtcpSsrc,\n        mediaSourceSsrc\n      );\n    }\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const exist = this.tracks.find((t) => {\n      if (t.rid) {\n        return t.rid === track.rid;\n      }\n      if (t.ssrc) {\n        return t.ssrc === track.ssrc;\n      }\n    });\n    if (exist) {\n      return false;\n    }\n    this.tracks.push(track);\n    if (track.ssrc) {\n      this.trackBySSRC[track.ssrc] = track;\n    }\n    if (track.rid) {\n      this.trackByRID[track.rid] = track;\n    }\n    return true;\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n\n    if (this.receiverTWCC) this.receiverTWCC.twccRunning = false;\n    this.nack.close();\n  }\n\n  async runRtcp() {\n    if (this.rtcpRunning || this.stopped) return;\n    this.rtcpRunning = true;\n\n    try {\n      while (this.rtcpRunning) {\n        await setTimeout(500 + Math.random() * 1000, undefined, {\n          signal: this.rtcpCancel.signal,\n        });\n\n        const reports = Object.entries(this.remoteStreams).map(\n          ([ssrc, stream]) => {\n            let lastSRtimestamp = 0,\n              delaySinceLastSR = 0;\n            if (this.lastSRtimestamp[ssrc]) {\n              lastSRtimestamp = this.lastSRtimestamp[ssrc];\n              const delaySeconds =\n                timestampSeconds() - this.receiveLastSRTimestamp[ssrc];\n              if (delaySeconds > 0 && delaySeconds < 65536) {\n                delaySinceLastSR = int(delaySeconds * 65536);\n              }\n            }\n\n            return new RtcpReceiverInfo({\n              ssrc: Number(ssrc),\n              fractionLost: stream.fraction_lost,\n              packetsLost: stream.packets_lost,\n              highestSequence: stream.max_seq,\n              jitter: stream.jitter,\n              lsr: lastSRtimestamp,\n              dlsr: delaySinceLastSR,\n            });\n          }\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\n          if (this.config.debug.receiverReportDelay) {\n            await setTimeout(this.config.debug.receiverReportDelay);\n          }\n          await this.dtlsTransport.sendRtcp([packet]);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  /**todo impl */\n  getStats() {}\n\n  async sendRtcpPLI(mediaSsrc: number) {\n    if (!this.pliEnabled) {\n      log(\"pli not supported\", { mediaSsrc });\n      return;\n    }\n\n    if (this.stopped) {\n      return;\n    }\n\n    log(\"sendRtcpPLI\", { mediaSsrc });\n\n    const packet = new RtcpPayloadSpecificFeedback({\n      feedback: new PictureLossIndication({\n        senderSsrc: this.rtcpSsrc,\n        mediaSsrc,\n      }),\n    });\n    try {\n      await this.dtlsTransport.sendRtcp([packet]);\n    } catch (error) {\n      log(error);\n    }\n  }\n\n  handleRtcpPacket(packet: RtcpPacket) {\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          const sr = packet as RtcpSrPacket;\n          this.lastSRtimestamp[sr.ssrc] = compactNtp(\n            sr.senderInfo.ntpTimestamp\n          );\n          this.receiveLastSRTimestamp[sr.ssrc] = timestampSeconds();\n\n          const track = this.trackBySSRC[packet.ssrc];\n          if (track) {\n            track.onReceiveRtcp.execute(packet);\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(packet);\n  }\n\n  handleRtpBySsrc = (packet: RtpPacket, extensions: Extensions) => {\n    const track = this.trackBySSRC[packet.header.ssrc];\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  handleRtpByRid = (packet: RtpPacket, rid: string, extensions: Extensions) => {\n    const track = this.trackByRID[rid];\n    if (!this.trackBySSRC[packet.header.ssrc]) {\n      this.trackBySSRC[packet.header.ssrc] = track;\n    }\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  private handleRTP(\n    packet: RtpPacket,\n    extensions: Extensions,\n    track?: MediaStreamTrack\n  ) {\n    if (this.stopped) {\n      return;\n    }\n\n    const codec = this.codecs[packet.header.payloadType];\n    if (!codec) {\n      // log(\"unknown codec \" + packet.header.payloadType);\n      return;\n    }\n\n    this.remoteStreams[packet.header.ssrc] =\n      this.remoteStreams[packet.header.ssrc] ??\n      new StreamStatistics(codec.clockRate);\n    this.remoteStreams[packet.header.ssrc].add(packet);\n\n    if (this.receiverTWCC) {\n      const transportSequenceNumber = extensions[\n        RTP_EXTENSION_URI.transportWideCC\n      ] as number;\n      if (!transportSequenceNumber == undefined) {\n        throw new Error(\"undefined\");\n      }\n\n      this.receiverTWCC.handleTWCC(transportSequenceNumber);\n    } else if (this.twccEnabled) {\n      this.setupTWCC(packet.header.ssrc);\n    }\n\n    if (codec.name.toLowerCase() === \"rtx\") {\n      const originalSsrc = this.ssrcByRtx[packet.header.ssrc];\n      const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n      const rtxCodec = this.codecs[codecParams[\"apt\"]];\n      if (packet.payload.length < 2) return;\n\n      packet = unwrapRtx(packet, rtxCodec.payloadType, originalSsrc);\n      track = this.trackBySSRC[originalSsrc];\n    }\n\n    let red: Red | undefined;\n    if (codec.name.toLowerCase() === \"red\") {\n      red = Red.deSerialize(packet.payload);\n      if (\n        !Object.keys(this.codecs).includes(\n          red.header.fields[0].blockPT.toString()\n        )\n      ) {\n        return;\n      }\n    }\n\n    if (track?.kind === \"video\" && this.nackEnabled) {\n      this.nack.addPacket(packet);\n    }\n\n    if (track) {\n      if (red) {\n        if (track.kind === \"audio\") {\n          const payloads = this.audioRedHandler.push(red, packet);\n          for (const packet of payloads) {\n            track.onReceiveRtp.execute(packet.clone());\n          }\n        } else {\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone());\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n\nexport function unwrapRtx(rtx: RtpPacket, payloadType: number, ssrc: number) {\n  const packet = new RtpPacket(\n    new RtpHeader({\n      payloadType,\n      marker: rtx.header.marker,\n      sequenceNumber: jspack.Unpack(\"!H\", rtx.payload.subarray(0, 2))[0],\n      timestamp: rtx.header.timestamp,\n      ssrc,\n    }),\n    rtx.payload.subarray(2)\n  );\n  return packet;\n}\n"]}
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{ debug } from \"debug\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\n\nimport { int } from \"../../../common/src\";\nimport {\n  PictureLossIndication,\n  Red,\n  RedHandler,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  RtpPacket,\n  unwrapRtx,\n} from \"../../../rtp/src\";\nimport { codecParametersFromString, PeerConfig, usePLI, useTWCC } from \"..\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, timestampSeconds } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport { RTCRtpCodecParameters, RTCRtpReceiveParameters } from \"./parameters\";\nimport { NackHandler } from \"./receiver/nack\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { StreamStatistics } from \"./receiver/statistics\";\nimport { Extensions } from \"./router\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpReceiver.ts\");\n\nexport class RTCRtpReceiver {\n  private readonly codecs: { [pt: number]: RTCRtpCodecParameters } = {};\n  private get codecArray() {\n    return Object.values(this.codecs).sort(\n      (a, b) => a.payloadType - b.payloadType\n    );\n  }\n  private readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new NackHandler(this);\n  private readonly audioRedHandler = new RedHandler();\n\n  readonly type = \"receiver\";\n  readonly uuid = uuid();\n  readonly tracks: MediaStreamTrack[] = [];\n  readonly trackBySSRC: { [ssrc: string]: MediaStreamTrack } = {};\n  readonly trackByRID: { [rid: string]: MediaStreamTrack } = {};\n  /**last sender Report Timestamp\n   * compactNtp\n   */\n  readonly lastSRtimestamp: { [ssrc: number]: number } = {};\n  /**seconds */\n  readonly receiveLastSRTimestamp: { [ssrc: number]: number } = {};\n  readonly onPacketLost = this.nack.onPacketLost;\n  readonly onRtcp = new Event<[RtcpPacket]>();\n\n  dtlsTransport!: RTCDtlsTransport;\n  sdesMid?: string;\n  latestRid?: string;\n  latestRepairedRid?: string;\n\n  receiverTWCC?: ReceiverTWCC;\n  stopped = false;\n  remoteStreamId?: string;\n  remoteTrackId?: string;\n\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n  private remoteStreams: { [ssrc: number]: StreamStatistics } = {};\n\n  constructor(\n    readonly config: PeerConfig,\n    public kind: Kind,\n    public rtcpSsrc: number\n  ) {}\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.dtlsTransport = dtls;\n  }\n\n  // todo fix\n  get track() {\n    return this.tracks[0];\n  }\n\n  get nackEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find((f) => f.type === \"nack\");\n  }\n\n  get twccEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === useTWCC().type\n    );\n  }\n\n  get pliEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === usePLI().type\n    );\n  }\n\n  prepareReceive(params: RTCRtpReceiveParameters) {\n    params.codecs.forEach((c) => {\n      this.codecs[c.payloadType] = c;\n    });\n    params.encodings.forEach((e) => {\n      if (e.rtx) {\n        this.ssrcByRtx[e.rtx.ssrc] = e.ssrc;\n      }\n    });\n  }\n\n  /**\n   * setup TWCC if supported\n   */\n  setupTWCC(mediaSourceSsrc: number) {\n    if (this.twccEnabled && !this.receiverTWCC) {\n      this.receiverTWCC = new ReceiverTWCC(\n        this.dtlsTransport,\n        this.rtcpSsrc,\n        mediaSourceSsrc\n      );\n    }\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const exist = this.tracks.find((t) => {\n      if (t.rid) {\n        return t.rid === track.rid;\n      }\n      if (t.ssrc) {\n        return t.ssrc === track.ssrc;\n      }\n    });\n    if (exist) {\n      return false;\n    }\n    this.tracks.push(track);\n    if (track.ssrc) {\n      this.trackBySSRC[track.ssrc] = track;\n    }\n    if (track.rid) {\n      this.trackByRID[track.rid] = track;\n    }\n    return true;\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n\n    if (this.receiverTWCC) this.receiverTWCC.twccRunning = false;\n    this.nack.close();\n  }\n\n  async runRtcp() {\n    if (this.rtcpRunning || this.stopped) return;\n    this.rtcpRunning = true;\n\n    try {\n      while (this.rtcpRunning) {\n        await setTimeout(500 + Math.random() * 1000, undefined, {\n          signal: this.rtcpCancel.signal,\n        });\n\n        const reports = Object.entries(this.remoteStreams).map(\n          ([ssrc, stream]) => {\n            let lastSRtimestamp = 0,\n              delaySinceLastSR = 0;\n            if (this.lastSRtimestamp[ssrc]) {\n              lastSRtimestamp = this.lastSRtimestamp[ssrc];\n              const delaySeconds =\n                timestampSeconds() - this.receiveLastSRTimestamp[ssrc];\n              if (delaySeconds > 0 && delaySeconds < 65536) {\n                delaySinceLastSR = int(delaySeconds * 65536);\n              }\n            }\n\n            return new RtcpReceiverInfo({\n              ssrc: Number(ssrc),\n              fractionLost: stream.fraction_lost,\n              packetsLost: stream.packets_lost,\n              highestSequence: stream.max_seq,\n              jitter: stream.jitter,\n              lsr: lastSRtimestamp,\n              dlsr: delaySinceLastSR,\n            });\n          }\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\n          if (this.config.debug.receiverReportDelay) {\n            await setTimeout(this.config.debug.receiverReportDelay);\n          }\n          await this.dtlsTransport.sendRtcp([packet]);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  /**todo impl */\n  getStats() {}\n\n  async sendRtcpPLI(mediaSsrc: number) {\n    if (!this.pliEnabled) {\n      log(\"pli not supported\", { mediaSsrc });\n      return;\n    }\n\n    if (this.stopped) {\n      return;\n    }\n\n    log(\"sendRtcpPLI\", { mediaSsrc });\n\n    const packet = new RtcpPayloadSpecificFeedback({\n      feedback: new PictureLossIndication({\n        senderSsrc: this.rtcpSsrc,\n        mediaSsrc,\n      }),\n    });\n    try {\n      await this.dtlsTransport.sendRtcp([packet]);\n    } catch (error) {\n      log(error);\n    }\n  }\n\n  handleRtcpPacket(packet: RtcpPacket) {\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          const sr = packet as RtcpSrPacket;\n          this.lastSRtimestamp[sr.ssrc] = compactNtp(\n            sr.senderInfo.ntpTimestamp\n          );\n          this.receiveLastSRTimestamp[sr.ssrc] = timestampSeconds();\n\n          const track = this.trackBySSRC[packet.ssrc];\n          if (track) {\n            track.onReceiveRtcp.execute(packet);\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(packet);\n  }\n\n  handleRtpBySsrc = (packet: RtpPacket, extensions: Extensions) => {\n    const track = this.trackBySSRC[packet.header.ssrc];\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  handleRtpByRid = (packet: RtpPacket, rid: string, extensions: Extensions) => {\n    const track = this.trackByRID[rid];\n    if (!this.trackBySSRC[packet.header.ssrc]) {\n      this.trackBySSRC[packet.header.ssrc] = track;\n    }\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  private handleRTP(\n    packet: RtpPacket,\n    extensions: Extensions,\n    track?: MediaStreamTrack\n  ) {\n    if (this.stopped) {\n      return;\n    }\n\n    const codec = this.codecs[packet.header.payloadType];\n    if (!codec) {\n      // log(\"unknown codec \" + packet.header.payloadType);\n      return;\n    }\n\n    this.remoteStreams[packet.header.ssrc] =\n      this.remoteStreams[packet.header.ssrc] ??\n      new StreamStatistics(codec.clockRate);\n    this.remoteStreams[packet.header.ssrc].add(packet);\n\n    if (this.receiverTWCC) {\n      const transportSequenceNumber = extensions[\n        RTP_EXTENSION_URI.transportWideCC\n      ] as number;\n      if (!transportSequenceNumber == undefined) {\n        throw new Error(\"undefined\");\n      }\n\n      this.receiverTWCC.handleTWCC(transportSequenceNumber);\n    } else if (this.twccEnabled) {\n      this.setupTWCC(packet.header.ssrc);\n    }\n\n    if (codec.name.toLowerCase() === \"rtx\") {\n      const originalSsrc = this.ssrcByRtx[packet.header.ssrc];\n      const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n      const rtxCodec = this.codecs[codecParams[\"apt\"]];\n      if (packet.payload.length < 2) return;\n\n      packet = unwrapRtx(packet, rtxCodec.payloadType, originalSsrc);\n      track = this.trackBySSRC[originalSsrc];\n    }\n\n    let red: Red | undefined;\n    if (codec.name.toLowerCase() === \"red\") {\n      red = Red.deSerialize(packet.payload);\n      if (\n        !Object.keys(this.codecs).includes(\n          red.header.fields[0].blockPT.toString()\n        )\n      ) {\n        return;\n      }\n    }\n\n    if (track?.kind === \"video\" && this.nackEnabled) {\n      this.nack.addPacket(packet);\n    }\n\n    if (track) {\n      if (red) {\n        if (track.kind === \"audio\") {\n          const payloads = this.audioRedHandler.push(red, packet);\n          for (const packet of payloads) {\n            track.onReceiveRtp.execute(packet.clone());\n          }\n        } else {\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone());\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n"]}

package/lib/webrtc/src/media/rtpSender.d.ts

@@ -84,4 +84,3 @@ export declare class RTCRtpSender {
     sendRtp(rtp: Buffer | RtpPacket): Promise<void>;
     handleRtcpPacket(rtcpPacket: RtcpPacket): void;
 }
-export declare function wrapRtx(packet: RtpPacket, payloadType: number, sequenceNumber: number, ssrc: number): RtpPacket;

package/lib/webrtc/src/media/rtpSender.js

@@ -47,7 +47,7 @@ var __importDefault = (this && this.__importDefault) || function (mod) {
     return (mod && mod.__esModule) ? mod : { "default": mod };
 };
 Object.defineProperty(exports, "__esModule", { value: true });
-exports.wrapRtx = exports.RTCRtpSender = void 0;
+exports.RTCRtpSender = void 0;
 const crypto_1 = require("crypto");
 const debug_1 = __importDefault(require("debug"));
 const jspack_1 = require("jspack");
@@ -600,7 +600,7 @@ class RTCRtpSender {
                                     }
                                     if (packet) {
                                         if (this.rtxPayloadType != undefined) {
-                                            packet = wrapRtx(packet, this.rtxPayloadType, this.rtxSequenceNumber, this.rtxSsrc);
+                                            packet = (0, src_2.wrapRtx)(packet, this.rtxPayloadType, this.rtxSequenceNumber, this.rtxSsrc);
                                             this.rtxSequenceNumber = (0, src_1.uint16Add)(this.rtxSequenceNumber, 1);
                                         }
                                         await this.dtlsTransport.sendRtp(packet.payload, packet.header);
@@ -635,20 +635,4 @@ class RTCRtpSender {
     }
 }
 exports.RTCRtpSender = RTCRtpSender;
-function wrapRtx(packet, payloadType, sequenceNumber, ssrc) {
-    const rtx = new src_2.RtpPacket(new src_2.RtpHeader({
-        payloadType,
-        marker: packet.header.marker,
-        sequenceNumber,
-        timestamp: packet.header.timestamp,
-        ssrc,
-        csrc: packet.header.csrc,
-        extensions: packet.header.extensions,
-    }), Buffer.concat([
-        Buffer.from(jspack_1.jspack.Pack("!H", [packet.header.sequenceNumber])),
-        packet.payload,
-    ]));
-    return rtx;
-}
-exports.wrapRtx = wrapRtx;
 //# sourceMappingURL=rtpSender.js.map

package/lib/webrtc/src/media/rtpSender.js.map

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  [10 Nov 1995 11:33:25.125 UTC]       [10 Nov 1995 11:33:36.5 UTC]\n   n                 SR(n)              A=b710:8000 (46864.500 s)\n   ---------------------------------------------------------------->\n                      v                 ^\n   ntp_sec =0xb44db705 v               ^ dlsr=0x0005:4000 (    5.250s)\n   ntp_frac=0x20000000  v             ^  lsr =0xb705:2000 (46853.125s)\n     (3024992005.125 s)  v           ^\n   r                      v         ^ RR(n)\n   ---------------------------------------------------------------->\n                          |<-DLSR->|\n                           (5.250 s)\n        \n   A     0xb710:8000 (46864.500 s)\n   DLSR -0x0005:4000 (    5.250 s)\n   LSR  -0xb705:2000 (46853.125 s)\n   -------------------------------\n   delay 0x0006:2000 (    6.125 s)\n        \nFigure 2: Example for round-trip time computation\n */\n\nimport { randomBytes } from \"crypto\";\nimport debug from \"debug\";\nimport { jspack } from \"jspack\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport * as uuid from \"uuid\";\n\nimport {\n  bufferWriter,\n  random16,\n  uint16Add,\n  uint32Add,\n} from \"../../../common/src\";\nimport {\n  Extension,\n  GenericNack,\n  PictureLossIndication,\n  ReceiverEstimatedMaxBitrate,\n  RedEncoder,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSenderInfo,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  RtpHeader,\n  RtpPacket,\n  SourceDescriptionChunk,\n  SourceDescriptionItem,\n  TransportWideCC,\n} from \"../../../rtp/src\";\nimport { codecParametersFromString } from \"..\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, milliTime, ntpTime, timestampSeconds } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport {\n  RTCRtpCodecParameters,\n  RTCRtpHeaderExtensionParameters,\n  RTCRtpSendParameters,\n} from \"./parameters\";\nimport { SenderBandwidthEstimator, SentInfo } from \"./sender/senderBWE\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpSender.ts\");\n\nconst RTP_HISTORY_SIZE = 128;\nconst RTT_ALPHA = 0.85;\n\nexport class RTCRtpSender {\n  readonly type = \"sender\";\n  readonly kind: Kind;\n  readonly ssrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n  readonly rtxSsrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n  streamId = uuid.v4();\n  readonly trackId = uuid.v4();\n  readonly onReady = new Event();\n  readonly onRtcp = new Event<[RtcpPacket]>();\n  readonly onPictureLossIndication = new Event<[]>();\n  readonly onGenericNack = new Event<[GenericNack]>();\n  readonly senderBWE = new SenderBandwidthEstimator();\n\n  private cname?: string;\n  private mid?: string;\n  private rtpStreamId?: string;\n  private repairedRtpStreamId?: string;\n  private rtxPayloadType?: number;\n  private rtxSequenceNumber = random16();\n  redRedundantPayloadType?: number;\n  private _redDistance = 2;\n  redEncoder = new RedEncoder(this._redDistance);\n  private headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n  private disposeTrack?: () => void;\n\n  // # stats\n  private lastSRtimestamp?: number;\n  private lastSentSRTimestamp?: number;\n  private ntpTimestamp = 0n;\n  private rtpTimestamp = 0;\n  private octetCount = 0;\n  private packetCount = 0;\n  private rtt?: number;\n  receiverEstimatedMaxBitrate: bigint = 0n;\n\n  // rtp\n  private sequenceNumber?: number;\n  private timestamp?: number;\n  private timestampOffset = 0;\n  private seqOffset = 0;\n  private rtpCache: RtpPacket[] = [];\n  codec?: RTCRtpCodecParameters;\n  public dtlsTransport!: RTCDtlsTransport;\n  private dtlsDisposer: (() => void)[] = [];\n\n  track?: MediaStreamTrack;\n  stopped = false;\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n\n  constructor(public trackOrKind: Kind | MediaStreamTrack) {\n    this.kind =\n      typeof this.trackOrKind === \"string\"\n        ? this.trackOrKind\n        : this.trackOrKind.kind;\n    if (trackOrKind instanceof MediaStreamTrack) {\n      if (trackOrKind.streamId) {\n        this.streamId = trackOrKind.streamId;\n      }\n      this.registerTrack(trackOrKind);\n    }\n  }\n\n  setDtlsTransport(dtlsTransport: RTCDtlsTransport) {\n    if (this.dtlsTransport) {\n      this.dtlsDisposer.forEach((dispose) => dispose());\n    }\n\n    this.dtlsTransport = dtlsTransport;\n    this.dtlsDisposer = [\n      this.dtlsTransport.onStateChange.subscribe((state) => {\n        if (state === \"connected\") {\n          this.onReady.execute();\n        }\n      }).unSubscribe,\n    ];\n  }\n\n  get redDistance() {\n    return this._redDistance;\n  }\n  set redDistance(n: number) {\n    this._redDistance = n;\n    this.redEncoder.distance = n;\n  }\n\n  prepareSend(params: RTCRtpSendParameters) {\n    this.cname = params.rtcp?.cname;\n    this.mid = params.muxId;\n    this.headerExtensions = params.headerExtensions;\n    this.rtpStreamId = params.rtpStreamId;\n    this.repairedRtpStreamId = params.repairedRtpStreamId;\n\n    this.codec = params.codecs[0];\n    if (this.track) {\n      this.track.codec = this.codec;\n    }\n\n    params.codecs.forEach((codec) => {\n      const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n      if (\n        codec.name.toLowerCase() === \"rtx\" &&\n        codecParams[\"apt\"] === this.codec?.payloadType\n      ) {\n        this.rtxPayloadType = codec.payloadType;\n      }\n      if (codec.name.toLowerCase() === \"red\") {\n        this.redRedundantPayloadType = Number(\n          (codec.parameters ?? \"\").split(\"/\")[0]\n        );\n      }\n    });\n  }\n\n  registerTrack(track: MediaStreamTrack) {\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    if (this.disposeTrack) {\n      this.disposeTrack();\n    }\n\n    track.id = this.trackId;\n\n    const { unSubscribe } = track.onReceiveRtp.subscribe(async (rtp) => {\n      await this.sendRtp(rtp);\n    });\n    this.track = track;\n    this.disposeTrack = unSubscribe;\n\n    if (this.codec) {\n      track.codec = this.codec;\n    }\n\n    track.onSourceChanged.subscribe((header) => {\n      this.replaceRTP(header);\n    });\n  }\n\n  async replaceTrack(track: MediaStreamTrack | null) {\n    if (track === null) {\n      // todo impl\n      return;\n    }\n\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    if (this.sequenceNumber != undefined) {\n      const header =\n        track.header || (await track.onReceiveRtp.asPromise())[0].header;\n\n      this.replaceRTP(header);\n    }\n\n    this.registerTrack(track);\n    log(\"replaceTrack\", \"ssrc\", track.ssrc, \"rid\", track.rid);\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n    if (this.disposeTrack) {\n      this.disposeTrack();\n    }\n    this.track = undefined;\n  }\n\n  async runRtcp() {\n    if (this.rtcpRunning || this.stopped) return;\n    this.rtcpRunning = true;\n\n    try {\n      while (this.rtcpRunning) {\n        await setTimeout(500 + Math.random() * 1000, undefined, {\n          signal: this.rtcpCancel.signal,\n        });\n\n        const packets: RtcpPacket[] = [\n          new RtcpSrPacket({\n            ssrc: this.ssrc,\n            senderInfo: new RtcpSenderInfo({\n              ntpTimestamp: this.ntpTimestamp,\n              rtpTimestamp: this.rtpTimestamp,\n              packetCount: this.packetCount,\n              octetCount: this.octetCount,\n            }),\n          }),\n        ];\n        this.lastSRtimestamp = compactNtp(this.ntpTimestamp);\n        this.lastSentSRTimestamp = timestampSeconds();\n\n        if (this.cname) {\n          packets.push(\n            new RtcpSourceDescriptionPacket({\n              chunks: [\n                new SourceDescriptionChunk({\n                  source: this.ssrc,\n                  items: [\n                    new SourceDescriptionItem({ type: 1, text: this.cname }),\n                  ],\n                }),\n              ],\n            })\n          );\n        }\n\n        try {\n          await this.dtlsTransport.sendRtcp(packets);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  replaceRTP(\n    {\n      sequenceNumber,\n      timestamp,\n    }: Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">,\n    discontinuity = false\n  ) {\n    if (this.sequenceNumber != undefined) {\n      this.seqOffset = uint16Add(this.sequenceNumber, -sequenceNumber);\n      if (discontinuity) {\n        this.seqOffset = uint16Add(this.seqOffset, 2);\n      }\n    }\n    if (this.timestamp != undefined) {\n      this.timestampOffset = uint32Add(this.timestamp, -timestamp);\n      if (discontinuity) {\n        this.timestampOffset = uint16Add(this.timestampOffset, 1);\n      }\n    }\n    this.rtpCache = [];\n    log(\"replaceRTP\", this.sequenceNumber, sequenceNumber, this.seqOffset);\n  }\n\n  async sendRtp(rtp: Buffer | RtpPacket) {\n    if (this.dtlsTransport.state !== \"connected\" || !this.codec) {\n      return;\n    }\n\n    rtp = Buffer.isBuffer(rtp) ? RtpPacket.deSerialize(rtp) : rtp;\n\n    const { header, payload } = rtp;\n    header.ssrc = this.ssrc;\n    header.payloadType = this.codec.payloadType;\n    header.timestamp = uint32Add(header.timestamp, this.timestampOffset);\n    header.sequenceNumber = uint16Add(header.sequenceNumber, this.seqOffset);\n    this.timestamp = header.timestamp;\n    this.sequenceNumber = header.sequenceNumber;\n\n    const ntptime = ntpTime();\n\n    header.extensions = this.headerExtensions\n      .map((extension) => {\n        const payload = (() => {\n          switch (extension.uri) {\n            case RTP_EXTENSION_URI.sdesMid:\n              if (this.mid) {\n                return Buffer.from(this.mid);\n              }\n              return;\n            // todo : sender simulcast unsupported now\n            case RTP_EXTENSION_URI.sdesRTPStreamID:\n              if (this.rtpStreamId) {\n                return Buffer.from(this.rtpStreamId);\n              }\n              return;\n            // todo : sender simulcast unsupported now\n            case RTP_EXTENSION_URI.repairedRtpStreamId:\n              if (this.repairedRtpStreamId) {\n                return Buffer.from(this.repairedRtpStreamId);\n              }\n              return;\n            case RTP_EXTENSION_URI.transportWideCC:\n              this.dtlsTransport.transportSequenceNumber = uint16Add(\n                this.dtlsTransport.transportSequenceNumber,\n                1\n              );\n              return bufferWriter(\n                [2],\n                [this.dtlsTransport.transportSequenceNumber]\n              );\n            case RTP_EXTENSION_URI.absSendTime:\n              const buf = Buffer.alloc(3);\n              const time = (ntptime >> 14n) & 0x00ffffffn;\n              buf.writeUIntBE(Number(time), 0, 3);\n              return buf;\n          }\n        })();\n\n        if (payload) return { id: extension.id, payload };\n      })\n      .filter((v) => v) as Extension[];\n\n    this.ntpTimestamp = ntptime;\n    this.rtpTimestamp = header.timestamp;\n    this.octetCount += payload.length;\n    this.packetCount = uint32Add(this.packetCount, 1);\n\n    this.rtpCache[header.sequenceNumber % RTP_HISTORY_SIZE] = rtp;\n\n    let rtpPayload = payload;\n\n    if (this.redRedundantPayloadType) {\n      this.redEncoder.push({\n        block: rtpPayload,\n        timestamp: header.timestamp,\n        blockPT: this.redRedundantPayloadType,\n      });\n      const red = this.redEncoder.build();\n      rtpPayload = red.serialize();\n    }\n\n    const size = await this.dtlsTransport.sendRtp(rtpPayload, header);\n\n    this.runRtcp();\n    const millitime = milliTime();\n    const sentInfo: SentInfo = {\n      wideSeq: this.dtlsTransport.transportSequenceNumber,\n      size,\n      sendingAtMs: millitime,\n      sentAtMs: millitime,\n    };\n    this.senderBWE.rtpPacketSent(sentInfo);\n  }\n\n  handleRtcpPacket(rtcpPacket: RtcpPacket) {\n    switch (rtcpPacket.type) {\n      case RtcpSrPacket.type:\n      case RtcpRrPacket.type:\n        {\n          const packet = rtcpPacket as RtcpSrPacket | RtcpRrPacket;\n          packet.reports\n            .filter((report) => report.ssrc === this.ssrc)\n            .forEach((report) => {\n              if (this.lastSRtimestamp === report.lsr && report.dlsr) {\n                if (this.lastSentSRTimestamp) {\n                  const rtt =\n                    timestampSeconds() -\n                    this.lastSentSRTimestamp -\n                    report.dlsr / 65536;\n                  if (this.rtt === undefined) {\n                    this.rtt = rtt;\n                  } else {\n                    this.rtt = RTT_ALPHA * this.rtt + (1 - RTT_ALPHA) * rtt;\n                  }\n                }\n              }\n            });\n        }\n        break;\n      case RtcpTransportLayerFeedback.type:\n        {\n          const packet = rtcpPacket as RtcpTransportLayerFeedback;\n          switch (packet.feedback.count) {\n            case TransportWideCC.count:\n              {\n                const feedback = packet.feedback as TransportWideCC;\n                this.senderBWE.receiveTWCC(feedback);\n              }\n              break;\n            case GenericNack.count:\n              {\n                const feedback = packet.feedback as GenericNack;\n                feedback.lost.forEach(async (seqNum) => {\n                  let packet: RtpPacket | undefined =\n                    this.rtpCache[seqNum % RTP_HISTORY_SIZE];\n                  if (packet && packet.header.sequenceNumber !== seqNum) {\n                    packet = undefined;\n                  }\n                  if (packet) {\n                    if (this.rtxPayloadType != undefined) {\n                      packet = wrapRtx(\n                        packet,\n                        this.rtxPayloadType,\n                        this.rtxSequenceNumber,\n                        this.rtxSsrc\n                      );\n                      this.rtxSequenceNumber = uint16Add(\n                        this.rtxSequenceNumber,\n                        1\n                      );\n                    }\n                    await this.dtlsTransport.sendRtp(\n                      packet.payload,\n                      packet.header\n                    );\n                  }\n                });\n                this.onGenericNack.execute(feedback);\n              }\n              break;\n          }\n        }\n        break;\n      case RtcpPayloadSpecificFeedback.type:\n        {\n          const packet = rtcpPacket as RtcpPayloadSpecificFeedback;\n          switch (packet.feedback.count) {\n            case ReceiverEstimatedMaxBitrate.count:\n              {\n                const feedback = packet.feedback as ReceiverEstimatedMaxBitrate;\n                this.receiverEstimatedMaxBitrate = feedback.bitrate;\n              }\n              break;\n            case PictureLossIndication.count:\n              {\n                this.onPictureLossIndication.execute();\n              }\n              break;\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(rtcpPacket);\n  }\n}\n\nexport function wrapRtx(\n  packet: RtpPacket,\n  payloadType: number,\n  sequenceNumber: number,\n  ssrc: number\n) {\n  const rtx = new RtpPacket(\n    new RtpHeader({\n      payloadType,\n      marker: packet.header.marker,\n      sequenceNumber,\n      timestamp: packet.header.timestamp,\n      ssrc,\n      csrc: packet.header.csrc,\n      extensions: packet.header.extensions,\n    }),\n    Buffer.concat([\n      Buffer.from(jspack.Pack(\"!H\", [packet.header.sequenceNumber])),\n      packet.payload,\n    ])\n  );\n  return rtx;\n}\n"]}
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  [10 Nov 1995 11:33:25.125 UTC]       [10 Nov 1995 11:33:36.5 UTC]\n   n                 SR(n)              A=b710:8000 (46864.500 s)\n   ---------------------------------------------------------------->\n                      v                 ^\n   ntp_sec =0xb44db705 v               ^ dlsr=0x0005:4000 (    5.250s)\n   ntp_frac=0x20000000  v             ^  lsr =0xb705:2000 (46853.125s)\n     (3024992005.125 s)  v           ^\n   r                      v         ^ RR(n)\n   ---------------------------------------------------------------->\n                          |<-DLSR->|\n                           (5.250 s)\n        \n   A     0xb710:8000 (46864.500 s)\n   DLSR -0x0005:4000 (    5.250 s)\n   LSR  -0xb705:2000 (46853.125 s)\n   -------------------------------\n   delay 0x0006:2000 (    6.125 s)\n        \nFigure 2: Example for round-trip time computation\n */\n\nimport { randomBytes } from \"crypto\";\nimport debug from \"debug\";\nimport { jspack } from \"jspack\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport * as uuid from \"uuid\";\n\nimport {\n  bufferWriter,\n  random16,\n  uint16Add,\n  uint32Add,\n} from \"../../../common/src\";\nimport {\n  Extension,\n  GenericNack,\n  PictureLossIndication,\n  ReceiverEstimatedMaxBitrate,\n  RedEncoder,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSenderInfo,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  RtpHeader,\n  RtpPacket,\n  SourceDescriptionChunk,\n  SourceDescriptionItem,\n  TransportWideCC,\n  wrapRtx,\n} from \"../../../rtp/src\";\nimport { codecParametersFromString } from \"..\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, milliTime, ntpTime, timestampSeconds } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport {\n  RTCRtpCodecParameters,\n  RTCRtpHeaderExtensionParameters,\n  RTCRtpSendParameters,\n} from \"./parameters\";\nimport { SenderBandwidthEstimator, SentInfo } from \"./sender/senderBWE\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpSender.ts\");\n\nconst RTP_HISTORY_SIZE = 128;\nconst RTT_ALPHA = 0.85;\n\nexport class RTCRtpSender {\n  readonly type = \"sender\";\n  readonly kind: Kind;\n  readonly ssrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n  readonly rtxSsrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n  streamId = uuid.v4();\n  readonly trackId = uuid.v4();\n  readonly onReady = new Event();\n  readonly onRtcp = new Event<[RtcpPacket]>();\n  readonly onPictureLossIndication = new Event<[]>();\n  readonly onGenericNack = new Event<[GenericNack]>();\n  readonly senderBWE = new SenderBandwidthEstimator();\n\n  private cname?: string;\n  private mid?: string;\n  private rtpStreamId?: string;\n  private repairedRtpStreamId?: string;\n  private rtxPayloadType?: number;\n  private rtxSequenceNumber = random16();\n  redRedundantPayloadType?: number;\n  private _redDistance = 2;\n  redEncoder = new RedEncoder(this._redDistance);\n  private headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n  private disposeTrack?: () => void;\n\n  // # stats\n  private lastSRtimestamp?: number;\n  private lastSentSRTimestamp?: number;\n  private ntpTimestamp = 0n;\n  private rtpTimestamp = 0;\n  private octetCount = 0;\n  private packetCount = 0;\n  private rtt?: number;\n  receiverEstimatedMaxBitrate: bigint = 0n;\n\n  // rtp\n  private sequenceNumber?: number;\n  private timestamp?: number;\n  private timestampOffset = 0;\n  private seqOffset = 0;\n  private rtpCache: RtpPacket[] = [];\n  codec?: RTCRtpCodecParameters;\n  public dtlsTransport!: RTCDtlsTransport;\n  private dtlsDisposer: (() => void)[] = [];\n\n  track?: MediaStreamTrack;\n  stopped = false;\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n\n  constructor(public trackOrKind: Kind | MediaStreamTrack) {\n    this.kind =\n      typeof this.trackOrKind === \"string\"\n        ? this.trackOrKind\n        : this.trackOrKind.kind;\n    if (trackOrKind instanceof MediaStreamTrack) {\n      if (trackOrKind.streamId) {\n        this.streamId = trackOrKind.streamId;\n      }\n      this.registerTrack(trackOrKind);\n    }\n  }\n\n  setDtlsTransport(dtlsTransport: RTCDtlsTransport) {\n    if (this.dtlsTransport) {\n      this.dtlsDisposer.forEach((dispose) => dispose());\n    }\n\n    this.dtlsTransport = dtlsTransport;\n    this.dtlsDisposer = [\n      this.dtlsTransport.onStateChange.subscribe((state) => {\n        if (state === \"connected\") {\n          this.onReady.execute();\n        }\n      }).unSubscribe,\n    ];\n  }\n\n  get redDistance() {\n    return this._redDistance;\n  }\n  set redDistance(n: number) {\n    this._redDistance = n;\n    this.redEncoder.distance = n;\n  }\n\n  prepareSend(params: RTCRtpSendParameters) {\n    this.cname = params.rtcp?.cname;\n    this.mid = params.muxId;\n    this.headerExtensions = params.headerExtensions;\n    this.rtpStreamId = params.rtpStreamId;\n    this.repairedRtpStreamId = params.repairedRtpStreamId;\n\n    this.codec = params.codecs[0];\n    if (this.track) {\n      this.track.codec = this.codec;\n    }\n\n    params.codecs.forEach((codec) => {\n      const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n      if (\n        codec.name.toLowerCase() === \"rtx\" &&\n        codecParams[\"apt\"] === this.codec?.payloadType\n      ) {\n        this.rtxPayloadType = codec.payloadType;\n      }\n      if (codec.name.toLowerCase() === \"red\") {\n        this.redRedundantPayloadType = Number(\n          (codec.parameters ?? \"\").split(\"/\")[0]\n        );\n      }\n    });\n  }\n\n  registerTrack(track: MediaStreamTrack) {\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    if (this.disposeTrack) {\n      this.disposeTrack();\n    }\n\n    track.id = this.trackId;\n\n    const { unSubscribe } = track.onReceiveRtp.subscribe(async (rtp) => {\n      await this.sendRtp(rtp);\n    });\n    this.track = track;\n    this.disposeTrack = unSubscribe;\n\n    if (this.codec) {\n      track.codec = this.codec;\n    }\n\n    track.onSourceChanged.subscribe((header) => {\n      this.replaceRTP(header);\n    });\n  }\n\n  async replaceTrack(track: MediaStreamTrack | null) {\n    if (track === null) {\n      // todo impl\n      return;\n    }\n\n    if (track.stopped) throw new Error(\"track is ended\");\n\n    if (this.sequenceNumber != undefined) {\n      const header =\n        track.header || (await track.onReceiveRtp.asPromise())[0].header;\n\n      this.replaceRTP(header);\n    }\n\n    this.registerTrack(track);\n    log(\"replaceTrack\", \"ssrc\", track.ssrc, \"rid\", track.rid);\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n    if (this.disposeTrack) {\n      this.disposeTrack();\n    }\n    this.track = undefined;\n  }\n\n  async runRtcp() {\n    if (this.rtcpRunning || this.stopped) return;\n    this.rtcpRunning = true;\n\n    try {\n      while (this.rtcpRunning) {\n        await setTimeout(500 + Math.random() * 1000, undefined, {\n          signal: this.rtcpCancel.signal,\n        });\n\n        const packets: RtcpPacket[] = [\n          new RtcpSrPacket({\n            ssrc: this.ssrc,\n            senderInfo: new RtcpSenderInfo({\n              ntpTimestamp: this.ntpTimestamp,\n              rtpTimestamp: this.rtpTimestamp,\n              packetCount: this.packetCount,\n              octetCount: this.octetCount,\n            }),\n          }),\n        ];\n        this.lastSRtimestamp = compactNtp(this.ntpTimestamp);\n        this.lastSentSRTimestamp = timestampSeconds();\n\n        if (this.cname) {\n          packets.push(\n            new RtcpSourceDescriptionPacket({\n              chunks: [\n                new SourceDescriptionChunk({\n                  source: this.ssrc,\n                  items: [\n                    new SourceDescriptionItem({ type: 1, text: this.cname }),\n                  ],\n                }),\n              ],\n            })\n          );\n        }\n\n        try {\n          await this.dtlsTransport.sendRtcp(packets);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  replaceRTP(\n    {\n      sequenceNumber,\n      timestamp,\n    }: Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">,\n    discontinuity = false\n  ) {\n    if (this.sequenceNumber != undefined) {\n      this.seqOffset = uint16Add(this.sequenceNumber, -sequenceNumber);\n      if (discontinuity) {\n        this.seqOffset = uint16Add(this.seqOffset, 2);\n      }\n    }\n    if (this.timestamp != undefined) {\n      this.timestampOffset = uint32Add(this.timestamp, -timestamp);\n      if (discontinuity) {\n        this.timestampOffset = uint16Add(this.timestampOffset, 1);\n      }\n    }\n    this.rtpCache = [];\n    log(\"replaceRTP\", this.sequenceNumber, sequenceNumber, this.seqOffset);\n  }\n\n  async sendRtp(rtp: Buffer | RtpPacket) {\n    if (this.dtlsTransport.state !== \"connected\" || !this.codec) {\n      return;\n    }\n\n    rtp = Buffer.isBuffer(rtp) ? RtpPacket.deSerialize(rtp) : rtp;\n\n    const { header, payload } = rtp;\n    header.ssrc = this.ssrc;\n    header.payloadType = this.codec.payloadType;\n    header.timestamp = uint32Add(header.timestamp, this.timestampOffset);\n    header.sequenceNumber = uint16Add(header.sequenceNumber, this.seqOffset);\n    this.timestamp = header.timestamp;\n    this.sequenceNumber = header.sequenceNumber;\n\n    const ntptime = ntpTime();\n\n    header.extensions = this.headerExtensions\n      .map((extension) => {\n        const payload = (() => {\n          switch (extension.uri) {\n            case RTP_EXTENSION_URI.sdesMid:\n              if (this.mid) {\n                return Buffer.from(this.mid);\n              }\n              return;\n            // todo : sender simulcast unsupported now\n            case RTP_EXTENSION_URI.sdesRTPStreamID:\n              if (this.rtpStreamId) {\n                return Buffer.from(this.rtpStreamId);\n              }\n              return;\n            // todo : sender simulcast unsupported now\n            case RTP_EXTENSION_URI.repairedRtpStreamId:\n              if (this.repairedRtpStreamId) {\n                return Buffer.from(this.repairedRtpStreamId);\n              }\n              return;\n            case RTP_EXTENSION_URI.transportWideCC:\n              this.dtlsTransport.transportSequenceNumber = uint16Add(\n                this.dtlsTransport.transportSequenceNumber,\n                1\n              );\n              return bufferWriter(\n                [2],\n                [this.dtlsTransport.transportSequenceNumber]\n              );\n            case RTP_EXTENSION_URI.absSendTime:\n              const buf = Buffer.alloc(3);\n              const time = (ntptime >> 14n) & 0x00ffffffn;\n              buf.writeUIntBE(Number(time), 0, 3);\n              return buf;\n          }\n        })();\n\n        if (payload) return { id: extension.id, payload };\n      })\n      .filter((v) => v) as Extension[];\n\n    this.ntpTimestamp = ntptime;\n    this.rtpTimestamp = header.timestamp;\n    this.octetCount += payload.length;\n    this.packetCount = uint32Add(this.packetCount, 1);\n\n    this.rtpCache[header.sequenceNumber % RTP_HISTORY_SIZE] = rtp;\n\n    let rtpPayload = payload;\n\n    if (this.redRedundantPayloadType) {\n      this.redEncoder.push({\n        block: rtpPayload,\n        timestamp: header.timestamp,\n        blockPT: this.redRedundantPayloadType,\n      });\n      const red = this.redEncoder.build();\n      rtpPayload = red.serialize();\n    }\n\n    const size = await this.dtlsTransport.sendRtp(rtpPayload, header);\n\n    this.runRtcp();\n    const millitime = milliTime();\n    const sentInfo: SentInfo = {\n      wideSeq: this.dtlsTransport.transportSequenceNumber,\n      size,\n      sendingAtMs: millitime,\n      sentAtMs: millitime,\n    };\n    this.senderBWE.rtpPacketSent(sentInfo);\n  }\n\n  handleRtcpPacket(rtcpPacket: RtcpPacket) {\n    switch (rtcpPacket.type) {\n      case RtcpSrPacket.type:\n      case RtcpRrPacket.type:\n        {\n          const packet = rtcpPacket as RtcpSrPacket | RtcpRrPacket;\n          packet.reports\n            .filter((report) => report.ssrc === this.ssrc)\n            .forEach((report) => {\n              if (this.lastSRtimestamp === report.lsr && report.dlsr) {\n                if (this.lastSentSRTimestamp) {\n                  const rtt =\n                    timestampSeconds() -\n                    this.lastSentSRTimestamp -\n                    report.dlsr / 65536;\n                  if (this.rtt === undefined) {\n                    this.rtt = rtt;\n                  } else {\n                    this.rtt = RTT_ALPHA * this.rtt + (1 - RTT_ALPHA) * rtt;\n                  }\n                }\n              }\n            });\n        }\n        break;\n      case RtcpTransportLayerFeedback.type:\n        {\n          const packet = rtcpPacket as RtcpTransportLayerFeedback;\n          switch (packet.feedback.count) {\n            case TransportWideCC.count:\n              {\n                const feedback = packet.feedback as TransportWideCC;\n                this.senderBWE.receiveTWCC(feedback);\n              }\n              break;\n            case GenericNack.count:\n              {\n                const feedback = packet.feedback as GenericNack;\n                feedback.lost.forEach(async (seqNum) => {\n                  let packet: RtpPacket | undefined =\n                    this.rtpCache[seqNum % RTP_HISTORY_SIZE];\n                  if (packet && packet.header.sequenceNumber !== seqNum) {\n                    packet = undefined;\n                  }\n                  if (packet) {\n                    if (this.rtxPayloadType != undefined) {\n                      packet = wrapRtx(\n                        packet,\n                        this.rtxPayloadType,\n                        this.rtxSequenceNumber,\n                        this.rtxSsrc\n                      );\n                      this.rtxSequenceNumber = uint16Add(\n                        this.rtxSequenceNumber,\n                        1\n                      );\n                    }\n                    await this.dtlsTransport.sendRtp(\n                      packet.payload,\n                      packet.header\n                    );\n                  }\n                });\n                this.onGenericNack.execute(feedback);\n              }\n              break;\n          }\n        }\n        break;\n      case RtcpPayloadSpecificFeedback.type:\n        {\n          const packet = rtcpPacket as RtcpPayloadSpecificFeedback;\n          switch (packet.feedback.count) {\n            case ReceiverEstimatedMaxBitrate.count:\n              {\n                const feedback = packet.feedback as ReceiverEstimatedMaxBitrate;\n                this.receiverEstimatedMaxBitrate = feedback.bitrate;\n              }\n              break;\n            case PictureLossIndication.count:\n              {\n                this.onPictureLossIndication.execute();\n              }\n              break;\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(rtcpPacket);\n  }\n}\n"]}

package/lib/webrtc/src/transport/dtls.js

@@ -255,7 +255,7 @@ class RTCDtlsTransport {
             }
             if (!(0, utils_1.isMedia)(data))
                 return;
-            if ((0, utils_1.isRtcp)(data)) {
+            if ((0, src_2.isRtcp)(data)) {
                 const dec = this.srtcp.decrypt(data);
                 const rtcps = src_2.RtcpPacketConverter.deSerialize(dec);
                 rtcps.forEach((rtcp) => this.router.routeRtcp(rtcp));

package/lib/webrtc/src/transport/dtls.js.map

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{ Certificate, PrivateKey } from \"@fidm/x509\";\nimport debug from \"debug\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 } from \"uuid\";\n\nimport {\n  DtlsClient,\n  DtlsServer,\n  DtlsSocket,\n  Transport,\n} from \"../../../dtls/src\";\nimport {\n  HashAlgorithm,\n  NamedCurveAlgorithm,\n  SignatureAlgorithm,\n  SignatureHash,\n} from \"../../../dtls/src/cipher/const\";\nimport { CipherContext } from \"../../../dtls/src/context/cipher\";\nimport { Profile } from \"../../../dtls/src/context/srtp\";\nimport { Connection } from \"../../../ice/src\";\nimport {\n  RtcpPacket,\n  RtcpPacketConverter,\n  RtpHeader,\n  RtpPacket,\n  SrtcpSession,\n  SrtpSession,\n} from \"../../../rtp/src\";\nimport { keyLength, saltLength } from \"../../../rtp/src/srtp/const\";\nimport { RtpRouter } from \"../media/router\";\nimport { PeerConfig } from \"../peerConnection\";\nimport { fingerprint, isDtls, isMedia, isRtcp } from \"../utils\";\nimport { RTCIceTransport } from \"./ice\";\n\nconst log = debug(\"werift:packages/webrtc/src/transport/dtls.ts\");\n\nexport class RTCDtlsTransport {\n  id = v4();\n  state: DtlsState = \"new\";\n  role: DtlsRole = \"auto\";\n  srtpStarted = false;\n  transportSequenceNumber = 0;\n\n  dataReceiver: (buf: Buffer) => void = () => {};\n  dtls?: DtlsSocket;\n  srtp!: SrtpSession;\n  srtcp!: SrtcpSession;\n\n  readonly onStateChange = new Event<[DtlsState]>();\n\n  localCertificate?: RTCCertificate;\n  private remoteParameters?: RTCDtlsParameters;\n\n  constructor(\n    readonly config: PeerConfig,\n    readonly iceTransport: RTCIceTransport,\n    readonly router: RtpRouter,\n    readonly certificates: RTCCertificate[],\n    private readonly srtpProfiles: Profile[] = []\n  ) {\n    this.localCertificate = this.certificates[0];\n  }\n\n  get localParameters() {\n    return new RTCDtlsParameters(\n      this.localCertificate ? this.localCertificate.getFingerprints() : [],\n      this.role\n    );\n  }\n\n  async setupCertificate() {\n    if (!this.localCertificate) {\n      const { certPem, keyPem, signatureHash } =\n        await CipherContext.createSelfSignedCertificateWithKey(\n          {\n            signature: SignatureAlgorithm.ecdsa_3,\n            hash: HashAlgorithm.sha256_4,\n          },\n          NamedCurveAlgorithm.secp256r1_23\n        );\n      this.localCertificate = new RTCCertificate(\n        keyPem,\n        certPem,\n        signatureHash\n      );\n    }\n    return this.localCertificate;\n  }\n\n  setRemoteParams(remoteParameters: RTCDtlsParameters) {\n    this.remoteParameters = remoteParameters;\n  }\n\n  async start() {\n    if (this.state !== \"new\") throw new Error();\n    if (this.remoteParameters?.fingerprints.length === 0) throw new Error();\n\n    if (this.role === \"auto\") {\n      if (this.iceTransport.role === \"controlling\") {\n        this.role = \"server\";\n      } else {\n        this.role = \"client\";\n      }\n    }\n\n    this.setState(\"connecting\");\n\n    await new Promise<void>(async (r) => {\n      if (this.role === \"server\") {\n        this.dtls = new DtlsServer({\n          cert: this.localCertificate?.certPem,\n          key: this.localCertificate?.privateKey,\n          signatureHash: this.localCertificate?.signatureHash,\n          transport: createIceTransport(this.iceTransport.connection),\n          srtpProfiles: this.srtpProfiles,\n          extendedMasterSecret: true,\n          // certificateRequest: true,\n        });\n      } else {\n        this.dtls = new DtlsClient({\n          cert: this.localCertificate?.certPem,\n          key: this.localCertificate?.privateKey,\n          signatureHash: this.localCertificate?.signatureHash,\n          transport: createIceTransport(this.iceTransport.connection),\n          srtpProfiles: this.srtpProfiles,\n          extendedMasterSecret: true,\n        });\n      }\n      this.dtls.onData.subscribe((buf) => {\n        if (\n          this.config.debug.inboundPacketLoss &&\n          this.config.debug.inboundPacketLoss / 100 < Math.random()\n        ) {\n          return;\n        }\n        this.dataReceiver(buf);\n      });\n      this.dtls.onClose.subscribe(() => {\n        this.setState(\"closed\");\n      });\n      this.dtls.onConnect.once(r);\n      this.dtls.onError.subscribe((error) => {\n        this.setState(\"failed\");\n        log(\"dtls failed\", error);\n      });\n\n      if (this.dtls instanceof DtlsClient) {\n        await setTimeout(100);\n        this.dtls.connect().catch((error) => {\n          this.setState(\"failed\");\n          log(\"dtls connect failed\", error);\n        });\n      }\n    });\n\n    if (this.srtpProfiles.length > 0) {\n      this.startSrtp();\n    }\n    this.dtls!.onConnect.subscribe(() => {\n      this.updateSrtpSession();\n      this.setState(\"connected\");\n    });\n    this.setState(\"connected\");\n\n    log(\"dtls connected\");\n  }\n\n  updateSrtpSession() {\n    if (!this.dtls) throw new Error();\n\n    const profile = this.dtls.srtp.srtpProfile;\n    if (!profile) {\n      throw new Error(\"need srtpProfile\");\n    }\n    log(\"selected SRTP Profile\", profile);\n\n    const { localKey, localSalt, remoteKey, remoteSalt } =\n      this.dtls.extractSessionKeys(keyLength(profile), saltLength(profile));\n\n    const config = {\n      keys: {\n        localMasterKey: localKey,\n        localMasterSalt: localSalt,\n        remoteMasterKey: remoteKey,\n        remoteMasterSalt: remoteSalt,\n      },\n      profile,\n    };\n    this.srtp = new SrtpSession(config);\n    this.srtcp = new SrtcpSession(config);\n  }\n\n  startSrtp() {\n    if (this.srtpStarted) return;\n    this.srtpStarted = true;\n\n    this.updateSrtpSession();\n\n    this.iceTransport.connection.onData.subscribe((data) => {\n      if (\n        this.config.debug.inboundPacketLoss &&\n        this.config.debug.inboundPacketLoss / 100 < Math.random()\n      ) {\n        return;\n      }\n\n      if (!isMedia(data)) return;\n      if (isRtcp(data)) {\n        const dec = this.srtcp.decrypt(data);\n        const rtcps = RtcpPacketConverter.deSerialize(dec);\n        rtcps.forEach((rtcp) => this.router.routeRtcp(rtcp));\n      } else {\n        const dec = this.srtp.decrypt(data);\n        const rtp = RtpPacket.deSerialize(dec);\n        try {\n          this.router.routeRtp(rtp);\n        } catch (error) {\n          log(\"router error\", error);\n        }\n      }\n    });\n  }\n\n  readonly sendData = async (data: Buffer) => {\n    if (\n      this.config.debug.outboundPacketLoss &&\n      this.config.debug.outboundPacketLoss / 100 < Math.random()\n    ) {\n      return;\n    }\n\n    if (!this.dtls) {\n      throw new Error(\"dtls not established\");\n    }\n    await this.dtls.send(data);\n  };\n\n  async sendRtp(payload: Buffer, header: RtpHeader): Promise<number> {\n    const enc = this.srtp.encrypt(payload, header);\n\n    if (\n      this.config.debug.outboundPacketLoss &&\n      this.config.debug.outboundPacketLoss / 100 < Math.random()\n    ) {\n      return enc.length;\n    }\n\n    await this.iceTransport.connection.send(enc).catch(() => {});\n    return enc.length;\n  }\n\n  async sendRtcp(packets: RtcpPacket[]) {\n    const payload = Buffer.concat(packets.map((packet) => packet.serialize()));\n    const enc = this.srtcp.encrypt(payload);\n\n    if (\n      this.config.debug.outboundPacketLoss &&\n      this.config.debug.outboundPacketLoss / 100 < Math.random()\n    ) {\n      return enc.length;\n    }\n\n    await this.iceTransport.connection.send(enc).catch(() => {});\n  }\n\n  private setState(state: DtlsState) {\n    if (state != this.state) {\n      this.state = state;\n      this.onStateChange.execute(state);\n    }\n  }\n\n  async stop() {\n    this.setState(\"closed\");\n    // todo impl send alert\n  }\n}\n\nexport const DtlsStates = [\n  \"new\",\n  \"connecting\",\n  \"connected\",\n  \"closed\",\n  \"failed\",\n] as const;\nexport type DtlsState = typeof DtlsStates[number];\n\nexport type DtlsRole = \"auto\" | \"server\" | \"client\";\n\nexport class RTCCertificate {\n  publicKey: string;\n  privateKey: string;\n\n  constructor(\n    privateKeyPem: string,\n    public certPem: string,\n    public signatureHash: SignatureHash\n  ) {\n    const cert = Certificate.fromPEM(Buffer.from(certPem));\n    this.publicKey = cert.publicKey.toPEM();\n    this.privateKey = PrivateKey.fromPEM(Buffer.from(privateKeyPem)).toPEM();\n  }\n\n  getFingerprints(): RTCDtlsFingerprint[] {\n    return [\n      new RTCDtlsFingerprint(\n        \"sha-256\",\n        fingerprint(\n          Certificate.fromPEM(Buffer.from(this.certPem)).raw,\n          \"sha256\"\n        )\n      ),\n    ];\n  }\n}\n\nexport type DtlsKeys = {\n  certPem: string;\n  keyPem: string;\n  signatureHash: SignatureHash;\n};\n\nexport class RTCDtlsFingerprint {\n  constructor(public algorithm: string, public value: string) {}\n}\n\nexport class RTCDtlsParameters {\n  constructor(\n    public fingerprints: RTCDtlsFingerprint[] = [],\n    public role: \"auto\" | \"client\" | \"server\"\n  ) {}\n}\n\nclass IceTransport implements Transport {\n  constructor(private ice: Connection) {\n    ice.onData.subscribe((buf) => {\n      if (isDtls(buf)) {\n        if (this.onData) this.onData(buf);\n      }\n    });\n  }\n  onData?: (buf: Buffer) => void;\n\n  readonly send = (data: Buffer) => {\n    return this.ice.send(data);\n  };\n\n  close() {\n    this.ice.close();\n  }\n}\n\nconst createIceTransport = (ice: Connection) => new IceTransport(ice);\n"]}
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{ Certificate, PrivateKey } from \"@fidm/x509\";\nimport debug from \"debug\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 } from \"uuid\";\n\nimport {\n  DtlsClient,\n  DtlsServer,\n  DtlsSocket,\n  Transport,\n} from \"../../../dtls/src\";\nimport {\n  HashAlgorithm,\n  NamedCurveAlgorithm,\n  SignatureAlgorithm,\n  SignatureHash,\n} from \"../../../dtls/src/cipher/const\";\nimport { CipherContext } from \"../../../dtls/src/context/cipher\";\nimport { Profile } from \"../../../dtls/src/context/srtp\";\nimport { Connection } from \"../../../ice/src\";\nimport {\n  isRtcp,\n  RtcpPacket,\n  RtcpPacketConverter,\n  RtpHeader,\n  RtpPacket,\n  SrtcpSession,\n  SrtpSession,\n} from \"../../../rtp/src\";\nimport { keyLength, saltLength } from \"../../../rtp/src/srtp/const\";\nimport { RtpRouter } from \"../media/router\";\nimport { PeerConfig } from \"../peerConnection\";\nimport { fingerprint, isDtls, isMedia } from \"../utils\";\nimport { RTCIceTransport } from \"./ice\";\n\nconst log = debug(\"werift:packages/webrtc/src/transport/dtls.ts\");\n\nexport class RTCDtlsTransport {\n  id = v4();\n  state: DtlsState = \"new\";\n  role: DtlsRole = \"auto\";\n  srtpStarted = false;\n  transportSequenceNumber = 0;\n\n  dataReceiver: (buf: Buffer) => void = () => {};\n  dtls?: DtlsSocket;\n  srtp!: SrtpSession;\n  srtcp!: SrtcpSession;\n\n  readonly onStateChange = new Event<[DtlsState]>();\n\n  localCertificate?: RTCCertificate;\n  private remoteParameters?: RTCDtlsParameters;\n\n  constructor(\n    readonly config: PeerConfig,\n    readonly iceTransport: RTCIceTransport,\n    readonly router: RtpRouter,\n    readonly certificates: RTCCertificate[],\n    private readonly srtpProfiles: Profile[] = []\n  ) {\n    this.localCertificate = this.certificates[0];\n  }\n\n  get localParameters() {\n    return new RTCDtlsParameters(\n      this.localCertificate ? this.localCertificate.getFingerprints() : [],\n      this.role\n    );\n  }\n\n  async setupCertificate() {\n    if (!this.localCertificate) {\n      const { certPem, keyPem, signatureHash } =\n        await CipherContext.createSelfSignedCertificateWithKey(\n          {\n            signature: SignatureAlgorithm.ecdsa_3,\n            hash: HashAlgorithm.sha256_4,\n          },\n          NamedCurveAlgorithm.secp256r1_23\n        );\n      this.localCertificate = new RTCCertificate(\n        keyPem,\n        certPem,\n        signatureHash\n      );\n    }\n    return this.localCertificate;\n  }\n\n  setRemoteParams(remoteParameters: RTCDtlsParameters) {\n    this.remoteParameters = remoteParameters;\n  }\n\n  async start() {\n    if (this.state !== \"new\") throw new Error();\n    if (this.remoteParameters?.fingerprints.length === 0) throw new Error();\n\n    if (this.role === \"auto\") {\n      if (this.iceTransport.role === \"controlling\") {\n        this.role = \"server\";\n      } else {\n        this.role = \"client\";\n      }\n    }\n\n    this.setState(\"connecting\");\n\n    await new Promise<void>(async (r) => {\n      if (this.role === \"server\") {\n        this.dtls = new DtlsServer({\n          cert: this.localCertificate?.certPem,\n          key: this.localCertificate?.privateKey,\n          signatureHash: this.localCertificate?.signatureHash,\n          transport: createIceTransport(this.iceTransport.connection),\n          srtpProfiles: this.srtpProfiles,\n          extendedMasterSecret: true,\n          // certificateRequest: true,\n        });\n      } else {\n        this.dtls = new DtlsClient({\n          cert: this.localCertificate?.certPem,\n          key: this.localCertificate?.privateKey,\n          signatureHash: this.localCertificate?.signatureHash,\n          transport: createIceTransport(this.iceTransport.connection),\n          srtpProfiles: this.srtpProfiles,\n          extendedMasterSecret: true,\n        });\n      }\n      this.dtls.onData.subscribe((buf) => {\n        if (\n          this.config.debug.inboundPacketLoss &&\n          this.config.debug.inboundPacketLoss / 100 < Math.random()\n        ) {\n          return;\n        }\n        this.dataReceiver(buf);\n      });\n      this.dtls.onClose.subscribe(() => {\n        this.setState(\"closed\");\n      });\n      this.dtls.onConnect.once(r);\n      this.dtls.onError.subscribe((error) => {\n        this.setState(\"failed\");\n        log(\"dtls failed\", error);\n      });\n\n      if (this.dtls instanceof DtlsClient) {\n        await setTimeout(100);\n        this.dtls.connect().catch((error) => {\n          this.setState(\"failed\");\n          log(\"dtls connect failed\", error);\n        });\n      }\n    });\n\n    if (this.srtpProfiles.length > 0) {\n      this.startSrtp();\n    }\n    this.dtls!.onConnect.subscribe(() => {\n      this.updateSrtpSession();\n      this.setState(\"connected\");\n    });\n    this.setState(\"connected\");\n\n    log(\"dtls connected\");\n  }\n\n  updateSrtpSession() {\n    if (!this.dtls) throw new Error();\n\n    const profile = this.dtls.srtp.srtpProfile;\n    if (!profile) {\n      throw new Error(\"need srtpProfile\");\n    }\n    log(\"selected SRTP Profile\", profile);\n\n    const { localKey, localSalt, remoteKey, remoteSalt } =\n      this.dtls.extractSessionKeys(keyLength(profile), saltLength(profile));\n\n    const config = {\n      keys: {\n        localMasterKey: localKey,\n        localMasterSalt: localSalt,\n        remoteMasterKey: remoteKey,\n        remoteMasterSalt: remoteSalt,\n      },\n      profile,\n    };\n    this.srtp = new SrtpSession(config);\n    this.srtcp = new SrtcpSession(config);\n  }\n\n  startSrtp() {\n    if (this.srtpStarted) return;\n    this.srtpStarted = true;\n\n    this.updateSrtpSession();\n\n    this.iceTransport.connection.onData.subscribe((data) => {\n      if (\n        this.config.debug.inboundPacketLoss &&\n        this.config.debug.inboundPacketLoss / 100 < Math.random()\n      ) {\n        return;\n      }\n\n      if (!isMedia(data)) return;\n      if (isRtcp(data)) {\n        const dec = this.srtcp.decrypt(data);\n        const rtcps = RtcpPacketConverter.deSerialize(dec);\n        rtcps.forEach((rtcp) => this.router.routeRtcp(rtcp));\n      } else {\n        const dec = this.srtp.decrypt(data);\n        const rtp = RtpPacket.deSerialize(dec);\n        try {\n          this.router.routeRtp(rtp);\n        } catch (error) {\n          log(\"router error\", error);\n        }\n      }\n    });\n  }\n\n  readonly sendData = async (data: Buffer) => {\n    if (\n      this.config.debug.outboundPacketLoss &&\n      this.config.debug.outboundPacketLoss / 100 < Math.random()\n    ) {\n      return;\n    }\n\n    if (!this.dtls) {\n      throw new Error(\"dtls not established\");\n    }\n    await this.dtls.send(data);\n  };\n\n  async sendRtp(payload: Buffer, header: RtpHeader): Promise<number> {\n    const enc = this.srtp.encrypt(payload, header);\n\n    if (\n      this.config.debug.outboundPacketLoss &&\n      this.config.debug.outboundPacketLoss / 100 < Math.random()\n    ) {\n      return enc.length;\n    }\n\n    await this.iceTransport.connection.send(enc).catch(() => {});\n    return enc.length;\n  }\n\n  async sendRtcp(packets: RtcpPacket[]) {\n    const payload = Buffer.concat(packets.map((packet) => packet.serialize()));\n    const enc = this.srtcp.encrypt(payload);\n\n    if (\n      this.config.debug.outboundPacketLoss &&\n      this.config.debug.outboundPacketLoss / 100 < Math.random()\n    ) {\n      return enc.length;\n    }\n\n    await this.iceTransport.connection.send(enc).catch(() => {});\n  }\n\n  private setState(state: DtlsState) {\n    if (state != this.state) {\n      this.state = state;\n      this.onStateChange.execute(state);\n    }\n  }\n\n  async stop() {\n    this.setState(\"closed\");\n    // todo impl send alert\n  }\n}\n\nexport const DtlsStates = [\n  \"new\",\n  \"connecting\",\n  \"connected\",\n  \"closed\",\n  \"failed\",\n] as const;\nexport type DtlsState = typeof DtlsStates[number];\n\nexport type DtlsRole = \"auto\" | \"server\" | \"client\";\n\nexport class RTCCertificate {\n  publicKey: string;\n  privateKey: string;\n\n  constructor(\n    privateKeyPem: string,\n    public certPem: string,\n    public signatureHash: SignatureHash\n  ) {\n    const cert = Certificate.fromPEM(Buffer.from(certPem));\n    this.publicKey = cert.publicKey.toPEM();\n    this.privateKey = PrivateKey.fromPEM(Buffer.from(privateKeyPem)).toPEM();\n  }\n\n  getFingerprints(): RTCDtlsFingerprint[] {\n    return [\n      new RTCDtlsFingerprint(\n        \"sha-256\",\n        fingerprint(\n          Certificate.fromPEM(Buffer.from(this.certPem)).raw,\n          \"sha256\"\n        )\n      ),\n    ];\n  }\n}\n\nexport type DtlsKeys = {\n  certPem: string;\n  keyPem: string;\n  signatureHash: SignatureHash;\n};\n\nexport class RTCDtlsFingerprint {\n  constructor(public algorithm: string, public value: string) {}\n}\n\nexport class RTCDtlsParameters {\n  constructor(\n    public fingerprints: RTCDtlsFingerprint[] = [],\n    public role: \"auto\" | \"client\" | \"server\"\n  ) {}\n}\n\nclass IceTransport implements Transport {\n  constructor(private ice: Connection) {\n    ice.onData.subscribe((buf) => {\n      if (isDtls(buf)) {\n        if (this.onData) this.onData(buf);\n      }\n    });\n  }\n  onData?: (buf: Buffer) => void;\n\n  readonly send = (data: Buffer) => {\n    return this.ice.send(data);\n  };\n\n  close() {\n    this.ice.close();\n  }\n}\n\nconst createIceTransport = (ice: Connection) => new IceTransport(ice);\n"]}

package/lib/webrtc/src/utils.d.ts

@@ -5,7 +5,6 @@ import { RTCIceServer } from "./peerConnection";
 export declare function fingerprint(file: Buffer, hashName: string): any;
 export declare function isDtls(buf: Buffer): boolean;
 export declare function isMedia(buf: Buffer): boolean;
-export declare function isRtcp(buf: Buffer): boolean;
 export declare function reverseSimulcastDirection(dir: "recv" | "send"): "recv" | "send";
 export declare const andDirection: (a: Direction, b: Direction) => "inactive" | "sendonly" | "recvonly" | "sendrecv";
 export declare function reverseDirection(dir: Direction): Direction;

package/lib/webrtc/src/utils.js

@@ -3,7 +3,7 @@ var __importDefault = (this && this.__importDefault) || function (mod) {
     return (mod && mod.__esModule) ? mod : { "default": mod };
 };
 Object.defineProperty(exports, "__esModule", { value: true });
-exports.createSelfSignedCertificate = exports.RtpBuilder = exports.parseIceServers = exports.compactNtp = exports.ntpTime = exports.timestampSeconds = exports.milliTime = exports.microTime = exports.reverseDirection = exports.andDirection = exports.reverseSimulcastDirection = exports.isRtcp = exports.isMedia = exports.isDtls = exports.fingerprint = void 0;
+exports.createSelfSignedCertificate = exports.RtpBuilder = exports.parseIceServers = exports.compactNtp = exports.ntpTime = exports.timestampSeconds = exports.milliTime = exports.microTime = exports.reverseDirection = exports.andDirection = exports.reverseSimulcastDirection = exports.isMedia = exports.isDtls = exports.fingerprint = void 0;
 /* eslint-disable prefer-const */
 const crypto_1 = require("crypto");
 const debug_1 = __importDefault(require("debug"));
@@ -31,10 +31,6 @@ function isMedia(buf) {
     return firstByte > 127 && firstByte < 192;
 }
 exports.isMedia = isMedia;
-function isRtcp(buf) {
-    return buf.length >= 2 && buf[1] >= 192 && buf[1] <= 208;
-}
-exports.isRtcp = isRtcp;
 function reverseSimulcastDirection(dir) {
     if (dir === "recv")
         return "send";

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eslint-disable prefer-const */\nimport { createHash } from \"crypto\";\nimport debug from \"debug\";\nimport { performance } from \"perf_hooks\";\n\nimport {\n  bufferReader,\n  bufferWriter,\n  random16,\n  random32,\n  uint16Add,\n  uint32Add,\n} from \"../../common/src\";\nimport { CipherContext } from \"../../dtls/src/context/cipher\";\nimport { Address } from \"../../ice/src\";\nimport { RtpHeader, RtpPacket } from \"../../rtp/src\";\nimport { Direction, Directions } from \"./media/rtpTransceiver\";\nimport { RTCIceServer } from \"./peerConnection\";\nconst now = require(\"nano-time\");\n\nconst log = debug(\"werift:packages/webrtc/src/utils.ts\");\n\nexport function fingerprint(file: Buffer, hashName: string) {\n  const upper = (s: string) => s.toUpperCase();\n  const colon = (s: any) => s.match(/(.{2})/g).join(\":\");\n\n  const hash = createHash(hashName).update(file).digest(\"hex\");\n\n  return colon(upper(hash));\n}\n\nexport function isDtls(buf: Buffer) {\n  const firstByte = buf[0];\n  return firstByte > 19 && firstByte < 64;\n}\n\nexport function isMedia(buf: Buffer) {\n  const firstByte = buf[0];\n  return firstByte > 127 && firstByte < 192;\n}\n\nexport function isRtcp(buf: Buffer) {\n  return buf.length >= 2 && buf[1] >= 192 && buf[1] <= 208;\n}\n\nexport function reverseSimulcastDirection(dir: \"recv\" | \"send\") {\n  if (dir === \"recv\") return \"send\";\n  return \"recv\";\n}\n\nexport const andDirection = (a: Direction, b: Direction) =>\n  Directions[Directions.indexOf(a) & Directions.indexOf(b)];\n\nexport function reverseDirection(dir: Direction): Direction {\n  if (dir === \"sendonly\") return \"recvonly\";\n  if (dir === \"recvonly\") return \"sendonly\";\n  return dir;\n}\n\nexport const microTime = () => now.micro() as number;\n\nexport const milliTime = () => new Date().getTime();\n\nexport const timestampSeconds = () => Date.now() / 1000;\n\n/**https://datatracker.ietf.org/doc/html/rfc3550#section-4 */\nexport const ntpTime = () => {\n  const now = performance.timeOrigin + performance.now() - Date.UTC(1900, 0, 1);\n\n  const seconds = now / 1000;\n  const [sec, msec] = seconds.toString().split(\".\").map(Number);\n\n  const buf = bufferWriter([4, 4], [sec, msec]);\n\n  return buf.readBigUInt64BE();\n};\n\n/**\n * https://datatracker.ietf.org/doc/html/rfc3550#section-4\n * @param ntp\n * @returns 32bit\n */\nexport const compactNtp = (ntp: bigint) => {\n  const buf = bufferWriter([8], [ntp]);\n  const [, sec, msec] = bufferReader(buf, [2, 2, 2, 2]);\n  return bufferWriter([2, 2], [sec, msec]).readUInt32BE();\n};\n\nexport function parseIceServers(iceServers: RTCIceServer[]) {\n  const url2Address = (url?: string) => {\n    if (!url) return;\n    const [address, port] = url.split(\":\");\n    return [address, parseInt(port)] as Address;\n  };\n\n  const stunServer = url2Address(\n    iceServers.find(({ urls }) => urls.includes(\"stun:\"))?.urls.slice(5)\n  );\n  const turnServer = url2Address(\n    iceServers.find(({ urls }) => urls.includes(\"turn:\"))?.urls.slice(5)\n  );\n  const { credential, username } =\n    iceServers.find(({ urls }) => urls.includes(\"turn:\")) || {};\n\n  const options = {\n    stunServer,\n    turnServer,\n    turnUsername: username,\n    turnPassword: credential,\n  };\n  log(\"iceOptions\", options);\n  return options;\n}\n\nexport class RtpBuilder {\n  sequenceNumber = random16();\n  timestamp = random32();\n\n  create(payload: Buffer) {\n    this.sequenceNumber = uint16Add(this.sequenceNumber, 1);\n    this.timestamp = uint32Add(this.timestamp, 960);\n\n    const header = new RtpHeader({\n      sequenceNumber: this.sequenceNumber,\n      timestamp: Number(this.timestamp),\n      payloadType: 96,\n      extension: true,\n      marker: false,\n      padding: false,\n    });\n    const rtp = new RtpPacket(header, payload);\n    return rtp;\n  }\n}\n\n/**\n *\n * @param signatureHash\n * @param namedCurveAlgorithm necessary when use ecdsa\n */\nexport const createSelfSignedCertificate =\n  CipherContext.createSelfSignedCertificateWithKey;\n"]}
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eslint-disable prefer-const */\nimport { createHash } from \"crypto\";\nimport debug from \"debug\";\nimport { performance } from \"perf_hooks\";\n\nimport {\n  bufferReader,\n  bufferWriter,\n  random16,\n  random32,\n  uint16Add,\n  uint32Add,\n} from \"../../common/src\";\nimport { CipherContext } from \"../../dtls/src/context/cipher\";\nimport { Address } from \"../../ice/src\";\nimport { RtpHeader, RtpPacket } from \"../../rtp/src\";\nimport { Direction, Directions } from \"./media/rtpTransceiver\";\nimport { RTCIceServer } from \"./peerConnection\";\nconst now = require(\"nano-time\");\n\nconst log = debug(\"werift:packages/webrtc/src/utils.ts\");\n\nexport function fingerprint(file: Buffer, hashName: string) {\n  const upper = (s: string) => s.toUpperCase();\n  const colon = (s: any) => s.match(/(.{2})/g).join(\":\");\n\n  const hash = createHash(hashName).update(file).digest(\"hex\");\n\n  return colon(upper(hash));\n}\n\nexport function isDtls(buf: Buffer) {\n  const firstByte = buf[0];\n  return firstByte > 19 && firstByte < 64;\n}\n\nexport function isMedia(buf: Buffer) {\n  const firstByte = buf[0];\n  return firstByte > 127 && firstByte < 192;\n}\n\nexport function reverseSimulcastDirection(dir: \"recv\" | \"send\") {\n  if (dir === \"recv\") return \"send\";\n  return \"recv\";\n}\n\nexport const andDirection = (a: Direction, b: Direction) =>\n  Directions[Directions.indexOf(a) & Directions.indexOf(b)];\n\nexport function reverseDirection(dir: Direction): Direction {\n  if (dir === \"sendonly\") return \"recvonly\";\n  if (dir === \"recvonly\") return \"sendonly\";\n  return dir;\n}\n\nexport const microTime = () => now.micro() as number;\n\nexport const milliTime = () => new Date().getTime();\n\nexport const timestampSeconds = () => Date.now() / 1000;\n\n/**https://datatracker.ietf.org/doc/html/rfc3550#section-4 */\nexport const ntpTime = () => {\n  const now = performance.timeOrigin + performance.now() - Date.UTC(1900, 0, 1);\n\n  const seconds = now / 1000;\n  const [sec, msec] = seconds.toString().split(\".\").map(Number);\n\n  const buf = bufferWriter([4, 4], [sec, msec]);\n\n  return buf.readBigUInt64BE();\n};\n\n/**\n * https://datatracker.ietf.org/doc/html/rfc3550#section-4\n * @param ntp\n * @returns 32bit\n */\nexport const compactNtp = (ntp: bigint) => {\n  const buf = bufferWriter([8], [ntp]);\n  const [, sec, msec] = bufferReader(buf, [2, 2, 2, 2]);\n  return bufferWriter([2, 2], [sec, msec]).readUInt32BE();\n};\n\nexport function parseIceServers(iceServers: RTCIceServer[]) {\n  const url2Address = (url?: string) => {\n    if (!url) return;\n    const [address, port] = url.split(\":\");\n    return [address, parseInt(port)] as Address;\n  };\n\n  const stunServer = url2Address(\n    iceServers.find(({ urls }) => urls.includes(\"stun:\"))?.urls.slice(5)\n  );\n  const turnServer = url2Address(\n    iceServers.find(({ urls }) => urls.includes(\"turn:\"))?.urls.slice(5)\n  );\n  const { credential, username } =\n    iceServers.find(({ urls }) => urls.includes(\"turn:\")) || {};\n\n  const options = {\n    stunServer,\n    turnServer,\n    turnUsername: username,\n    turnPassword: credential,\n  };\n  log(\"iceOptions\", options);\n  return options;\n}\n\nexport class RtpBuilder {\n  sequenceNumber = random16();\n  timestamp = random32();\n\n  create(payload: Buffer) {\n    this.sequenceNumber = uint16Add(this.sequenceNumber, 1);\n    this.timestamp = uint32Add(this.timestamp, 960);\n\n    const header = new RtpHeader({\n      sequenceNumber: this.sequenceNumber,\n      timestamp: Number(this.timestamp),\n      payloadType: 96,\n      extension: true,\n      marker: false,\n      padding: false,\n    });\n    const rtp = new RtpPacket(header, payload);\n    return rtp;\n  }\n}\n\n/**\n *\n * @param signatureHash\n * @param namedCurveAlgorithm necessary when use ecdsa\n */\nexport const createSelfSignedCertificate =\n  CipherContext.createSelfSignedCertificateWithKey;\n"]}

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "werift",
-  "version": "0.18.1",
+  "version": "0.18.2",
   "description": "WebRTC Implementation for TypeScript (Node.js)",
   "keywords": [
     "WebRTC",

package/src/media/receiver/nack.ts

@@ -121,7 +121,7 @@ export class NackHandler {
           mediaSourceSsrc: this.mediaSourceSsrc,
           lost: this.lostSeqNumbers,
         });
-        // log("sendNack", nack.toJSON());
+
         const rtcp = new RtcpTransportLayerFeedback({
           feedback: nack,
         });

package/src/media/rtpReceiver.ts

@@ -1,5 +1,4 @@
 import { debug } from "debug";
-import { jspack } from "jspack";
 import Event from "rx.mini";
 import { setTimeout } from "timers/promises";
 import { v4 as uuid } from "uuid";
@@ -14,8 +13,8 @@ import {
   RtcpReceiverInfo,
   RtcpRrPacket,
   RtcpSrPacket,
-  RtpHeader,
   RtpPacket,
+  unwrapRtx,
 } from "../../../rtp/src";
 import { codecParametersFromString, PeerConfig, usePLI, useTWCC } from "..";
 import { RTCDtlsTransport } from "../transport/dtls";
@@ -345,17 +344,3 @@ export class RTCRtpReceiver {
     this.runRtcp();
   }
 }
-
-export function unwrapRtx(rtx: RtpPacket, payloadType: number, ssrc: number) {
-  const packet = new RtpPacket(
-    new RtpHeader({
-      payloadType,
-      marker: rtx.header.marker,
-      sequenceNumber: jspack.Unpack("!H", rtx.payload.subarray(0, 2))[0],
-      timestamp: rtx.header.timestamp,
-      ssrc,
-    }),
-    rtx.payload.subarray(2)
-  );
-  return packet;
-}