werift 0.19.0 → 0.19.1

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

@@ -1 +1 @@
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debug from \"debug\";\nimport { jspack } from \"jspack\";\nimport { Event } from \"rx.mini\";\nimport * as uuid from \"uuid\";\n\nimport { SCTP, SCTP_STATE, Transport } from \"../../../sctp/src\";\nimport {\n  DATA_CHANNEL_ACK,\n  DATA_CHANNEL_OPEN,\n  DATA_CHANNEL_RELIABLE,\n  WEBRTC_BINARY,\n  WEBRTC_BINARY_EMPTY,\n  WEBRTC_DCEP,\n  WEBRTC_STRING,\n  WEBRTC_STRING_EMPTY,\n} from \"../const\";\nimport {\n  DCState,\n  RTCDataChannel,\n  RTCDataChannelParameters,\n} from \"../dataChannel\";\nimport { RTCDtlsTransport } from \"./dtls\";\n\nconst log = debug(\"werift:packages/webrtc/src/transport/sctp.ts\");\n\nexport class RTCSctpTransport {\n  dtlsTransport!: RTCDtlsTransport;\n  sctp!: SCTP;\n\n  readonly onDataChannel = new Event<[RTCDataChannel]>();\n  readonly id = uuid.v4();\n\n  mid?: string;\n  mLineIndex?: number;\n  bundled = false;\n  dataChannels: { [key: number]: RTCDataChannel } = {};\n\n  private dataChannelQueue: [RTCDataChannel, number, Buffer][] = [];\n  private dataChannelId?: number;\n  private eventDisposer: (() => void)[] = [];\n\n  constructor(public port = 5000) {}\n\n  setDtlsTransport(dtlsTransport: RTCDtlsTransport) {\n    if (this.dtlsTransport && this.dtlsTransport.id === dtlsTransport.id) {\n      return;\n    }\n\n    this.eventDisposer.forEach((dispose) => dispose());\n\n    this.dtlsTransport = dtlsTransport;\n    this.sctp = new SCTP(new BridgeDtls(this.dtlsTransport), this.port);\n\n    this.eventDisposer = [\n      ...[\n        this.sctp.onReceive.subscribe(this.datachannelReceive),\n        this.sctp.onReconfigStreams.subscribe((ids: number[]) => {\n          ids.forEach((id) => {\n            const dc = this.dataChannels[id];\n            if (!dc) return;\n            // todo fix\n            dc.setReadyState(\"closing\");\n            dc.setReadyState(\"closed\");\n            delete this.dataChannels[id];\n          });\n        }),\n        this.sctp.stateChanged.connected.subscribe(() => {\n          Object.values(this.dataChannels).forEach((channel) => {\n            if (channel.negotiated && channel.readyState !== \"open\") {\n              channel.setReadyState(\"open\");\n            }\n          });\n          this.dataChannelFlush();\n        }),\n        this.sctp.stateChanged.closed.subscribe(() => {\n          Object.values(this.dataChannels).forEach((dc) => {\n            dc.setReadyState(\"closed\");\n          });\n          this.dataChannels = {};\n        }),\n        this.dtlsTransport.onStateChange.subscribe((state) => {\n          if (state === \"closed\") {\n            this.sctp.setState(SCTP_STATE.CLOSED);\n          }\n        }),\n      ].map((e) => e.unSubscribe),\n      () => (this.sctp.onSackReceived = async () => {}),\n    ];\n\n    this.sctp.onSackReceived = async () => {\n      await this.dataChannelFlush();\n    };\n  }\n\n  private get isServer() {\n    return this.dtlsTransport.iceTransport.role !== \"controlling\";\n  }\n\n  channelByLabel(label: string) {\n    return Object.values(this.dataChannels).find((d) => d.label === label);\n  }\n\n  private datachannelReceive = async (\n    streamId: number,\n    ppId: number,\n    data: Buffer,\n  ) => {\n    if (ppId === WEBRTC_DCEP && data.length > 0) {\n      log(\"DCEP\", streamId, ppId, data);\n      switch (data[0]) {\n        case DATA_CHANNEL_OPEN:\n          {\n            if (data.length < 12) {\n              log(\"DATA_CHANNEL_OPEN data.length not enough\");\n              return;\n            }\n\n            if (!Object.keys(this.dataChannels).includes(streamId.toString())) {\n              const [\n                ,\n                channelType,\n                ,\n                reliability,\n                labelLength,\n                protocolLength,\n              ] = jspack.Unpack(\"!BBHLHH\", data);\n\n              let pos = 12;\n              const label = data.slice(pos, pos + labelLength).toString(\"utf8\");\n              pos += labelLength;\n              const protocol = data\n                .slice(pos, pos + protocolLength)\n                .toString(\"utf8\");\n\n              log(\"DATA_CHANNEL_OPEN\", {\n                channelType,\n                reliability,\n                streamId,\n                label,\n                protocol,\n              });\n\n              const maxRetransmits =\n                (channelType & 0x03) === 1 ? reliability : undefined;\n              const maxPacketLifeTime =\n                (channelType & 0x03) === 2 ? reliability : undefined;\n\n              // # register channel\n              const parameters = new RTCDataChannelParameters({\n                label,\n                ordered: (channelType & 0x80) === 0,\n                maxPacketLifeTime,\n                maxRetransmits,\n                protocol,\n                id: streamId,\n              });\n              const channel = new RTCDataChannel(this, parameters, false);\n              channel.isCreatedByRemote = true;\n              this.dataChannels[streamId] = channel;\n\n              this.onDataChannel.execute(channel);\n              channel.setReadyState(\"open\");\n            } else {\n              log(\"datachannel already opened\", \"retransmit ack\");\n            }\n\n            const channel = this.dataChannels[streamId];\n            this.dataChannelQueue.push([\n              channel,\n              WEBRTC_DCEP,\n              Buffer.from(jspack.Pack(\"!B\", [DATA_CHANNEL_ACK])),\n            ]);\n            await this.dataChannelFlush();\n          }\n          break;\n        case DATA_CHANNEL_ACK:\n          log(\"DATA_CHANNEL_ACK\", streamId, ppId);\n          const channel = this.dataChannels[streamId];\n          if (!channel) {\n            throw new Error(\"channel not found\");\n          }\n          channel.setReadyState(\"open\");\n          break;\n      }\n    } else {\n      const channel = this.dataChannels[streamId];\n      if (channel) {\n        const msg = (() => {\n          switch (ppId) {\n            case WEBRTC_STRING:\n              return data.toString(\"utf8\");\n            case WEBRTC_STRING_EMPTY:\n              return \"\";\n            case WEBRTC_BINARY:\n              return data;\n            case WEBRTC_BINARY_EMPTY:\n              return Buffer.from([]);\n            default:\n              throw new Error();\n          }\n        })();\n\n        channel.message.execute(msg);\n        channel.emit(\"message\", { data: msg });\n        if (channel.onmessage) {\n          channel.onmessage({ data: msg });\n        }\n      }\n    }\n  };\n\n  dataChannelAddNegotiated(channel: RTCDataChannel) {\n    if (channel.id == undefined) {\n      throw new Error();\n    }\n    if (this.dataChannels[channel.id]) {\n      throw new Error();\n    }\n\n    this.dataChannels[channel.id] = channel;\n\n    if (this.sctp.associationState === SCTP_STATE.ESTABLISHED) {\n      channel.setReadyState(\"open\");\n    }\n  }\n\n  dataChannelOpen(channel: RTCDataChannel) {\n    if (channel.id) {\n      if (this.dataChannels[channel.id])\n        throw new Error(\n          `Data channel with ID ${channel.id} already registered`,\n        );\n      this.dataChannels[channel.id] = channel;\n    }\n\n    let channelType = DATA_CHANNEL_RELIABLE;\n    const priority = 0;\n    let reliability = 0;\n\n    if (!channel.ordered) {\n      channelType = 0x80;\n    }\n    if (channel.maxRetransmits) {\n      channelType = 1;\n      reliability = channel.maxRetransmits;\n    } else if (channel.maxPacketLifeTime) {\n      channelType = 2;\n      reliability = channel.maxPacketLifeTime;\n    }\n\n    // 5.1.  DATA_CHANNEL_OPEN Message\n    const data = jspack.Pack(\"!BBHLHH\", [\n      DATA_CHANNEL_OPEN,\n      channelType,\n      priority,\n      reliability,\n      channel.label.length,\n      channel.protocol.length,\n    ]);\n    const send = Buffer.concat([\n      Buffer.from(data),\n      Buffer.from(channel.label, \"utf8\"),\n      Buffer.from(channel.protocol, \"utf8\"),\n    ]);\n    this.dataChannelQueue.push([channel, WEBRTC_DCEP, send]);\n    this.dataChannelFlush();\n  }\n\n  private async dataChannelFlush() {\n    // \"\"\"\n    // Try to flush buffered data to the SCTP layer.\n\n    // We wait until the association is established, as we need to know\n    // whether we are a client or a server to correctly assign an odd/even ID\n    // to the data channels.\n    // \"\"\"\n\n    if (this.sctp.associationState != SCTP_STATE.ESTABLISHED) return;\n    if (this.sctp.outboundQueue.length > 0) return;\n\n    while (this.dataChannelQueue.length > 0) {\n      const [channel, protocol, userData] = this.dataChannelQueue.shift()!;\n\n      let streamId = channel.id;\n      if (streamId === undefined) {\n        streamId = this.dataChannelId!;\n        while (Object.keys(this.dataChannels).includes(streamId.toString())) {\n          streamId += 2;\n        }\n        this.dataChannels[streamId] = channel;\n        channel.setId(streamId);\n      }\n\n      if (protocol === WEBRTC_DCEP) {\n        await this.sctp.send(streamId, protocol, userData, {\n          ordered: true,\n        });\n      } else {\n        const expiry = channel.maxPacketLifeTime\n          ? Date.now() + channel.maxPacketLifeTime / 1000\n          : undefined;\n\n        await this.sctp.send(streamId, protocol, userData, {\n          expiry,\n          maxRetransmits: channel.maxRetransmits,\n          ordered: channel.ordered,\n        });\n        channel.addBufferedAmount(-userData.length);\n      }\n    }\n  }\n\n  datachannelSend = (channel: RTCDataChannel, data: Buffer | string) => {\n    channel.addBufferedAmount(data.length);\n\n    this.dataChannelQueue.push(\n      typeof data === \"string\"\n        ? [channel, WEBRTC_STRING, Buffer.from(data)]\n        : [channel, WEBRTC_BINARY, data],\n    );\n\n    if (this.sctp.associationState !== SCTP_STATE.ESTABLISHED) {\n      log(\"sctp not established\", this.sctp.associationState);\n    }\n\n    this.dataChannelFlush();\n  };\n\n  static getCapabilities() {\n    return new RTCSctpCapabilities(65536);\n  }\n\n  setRemotePort(port: number) {\n    this.sctp.setRemotePort(port);\n  }\n\n  async start(remotePort: number) {\n    if (this.isServer) {\n      this.dataChannelId = 0;\n    } else {\n      this.dataChannelId = 1;\n    }\n    this.sctp.isServer = this.isServer;\n\n    await this.sctp.start(remotePort);\n  }\n\n  async stop() {\n    this.dtlsTransport.dataReceiver = () => {};\n    await this.sctp.stop();\n  }\n\n  dataChannelClose(channel: RTCDataChannel) {\n    if (!([\"closing\", \"closed\"] as DCState[]).includes(channel.readyState)) {\n      channel.setReadyState(\"closing\");\n\n      if (this.sctp.associationState === SCTP_STATE.ESTABLISHED) {\n        this.sctp.reconfigQueue.push(channel.id);\n        if (this.sctp.reconfigQueue.length === 1) {\n          this.sctp.transmitReconfigRequest();\n        }\n      } else {\n        this.dataChannelQueue = this.dataChannelQueue.filter(\n          (queueItem) => queueItem[0].id !== channel.id,\n        );\n        if (channel.id) {\n          delete this.dataChannels[channel.id];\n        }\n        channel.setReadyState(\"closed\");\n      }\n    }\n  }\n}\n\nexport class RTCSctpCapabilities {\n  constructor(public maxMessageSize: number) {}\n}\n\nclass BridgeDtls implements Transport {\n  constructor(private dtls: RTCDtlsTransport) {}\n  set onData(onData: (buf: Buffer) => void) {\n    this.dtls.dataReceiver = onData;\n  }\n  readonly send = (data: Buffer) => {\n    return this.dtls.sendData(data);\n  };\n  close() {}\n}\n"]}
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debug from \"debug\";\nimport { jspack } from \"jspack\";\nimport { Event } from \"rx.mini\";\nimport * as uuid from \"uuid\";\n\nimport { SCTP, SCTP_STATE, Transport } from \"../../../sctp/src\";\nimport {\n  DATA_CHANNEL_ACK,\n  DATA_CHANNEL_OPEN,\n  DATA_CHANNEL_RELIABLE,\n  WEBRTC_BINARY,\n  WEBRTC_BINARY_EMPTY,\n  WEBRTC_DCEP,\n  WEBRTC_STRING,\n  WEBRTC_STRING_EMPTY,\n} from \"../const\";\nimport {\n  DCState,\n  RTCDataChannel,\n  RTCDataChannelParameters,\n} from \"../dataChannel\";\nimport { RTCDtlsTransport } from \"./dtls\";\n\nconst log = debug(\"werift:packages/webrtc/src/transport/sctp.ts\");\n\nexport class RTCSctpTransport {\n  dtlsTransport!: RTCDtlsTransport;\n  sctp!: SCTP;\n\n  readonly onDataChannel = new Event<[RTCDataChannel]>();\n  readonly id = uuid.v4();\n\n  mid?: string;\n  mLineIndex?: number;\n  bundled = false;\n  dataChannels: { [key: number]: RTCDataChannel } = {};\n\n  private dataChannelQueue: [RTCDataChannel, number, Buffer][] = [];\n  private dataChannelId?: number;\n  private eventDisposer: (() => void)[] = [];\n\n  constructor(public port = 5000) {}\n\n  setDtlsTransport(dtlsTransport: RTCDtlsTransport) {\n    if (this.dtlsTransport && this.dtlsTransport.id === dtlsTransport.id) {\n      return;\n    }\n\n    this.eventDisposer.forEach((dispose) => dispose());\n\n    this.dtlsTransport = dtlsTransport;\n    this.sctp = new SCTP(new BridgeDtls(this.dtlsTransport), this.port);\n\n    this.eventDisposer = [\n      ...[\n        this.sctp.onReceive.subscribe(this.datachannelReceive),\n        this.sctp.onReconfigStreams.subscribe((ids: number[]) => {\n          ids.forEach((id) => {\n            const dc = this.dataChannels[id];\n            if (!dc) return;\n            // todo fix\n            dc.setReadyState(\"closing\");\n            dc.setReadyState(\"closed\");\n            delete this.dataChannels[id];\n          });\n        }),\n        this.sctp.stateChanged.connected.subscribe(() => {\n          Object.values(this.dataChannels).forEach((channel) => {\n            if (channel.negotiated && channel.readyState !== \"open\") {\n              channel.setReadyState(\"open\");\n            }\n          });\n          this.dataChannelFlush();\n        }),\n        this.sctp.stateChanged.closed.subscribe(() => {\n          Object.values(this.dataChannels).forEach((dc) => {\n            dc.setReadyState(\"closed\");\n          });\n          this.dataChannels = {};\n        }),\n        this.dtlsTransport.onStateChange.subscribe((state) => {\n          if (state === \"closed\") {\n            this.sctp.setState(SCTP_STATE.CLOSED);\n          }\n        }),\n      ].map((e) => e.unSubscribe),\n      () => (this.sctp.onSackReceived = async () => {}),\n    ];\n\n    this.sctp.onSackReceived = async () => {\n      await this.dataChannelFlush();\n    };\n  }\n\n  private get isServer() {\n    return this.dtlsTransport.iceTransport.role !== \"controlling\";\n  }\n\n  channelByLabel(label: string) {\n    return Object.values(this.dataChannels).find((d) => d.label === label);\n  }\n\n  private datachannelReceive = async (\n    streamId: number,\n    ppId: number,\n    data: Buffer,\n  ) => {\n    if (ppId === WEBRTC_DCEP && data.length > 0) {\n      log(\"DCEP\", streamId, ppId, data);\n      switch (data[0]) {\n        case DATA_CHANNEL_OPEN: {\n          if (data.length < 12) {\n            log(\"DATA_CHANNEL_OPEN data.length not enough\");\n            return;\n          }\n\n          if (!Object.keys(this.dataChannels).includes(streamId.toString())) {\n            const [, channelType, , reliability, labelLength, protocolLength] =\n              jspack.Unpack(\"!BBHLHH\", data);\n\n            let pos = 12;\n            const label = data.slice(pos, pos + labelLength).toString(\"utf8\");\n            pos += labelLength;\n            const protocol = data\n              .slice(pos, pos + protocolLength)\n              .toString(\"utf8\");\n\n            log(\"DATA_CHANNEL_OPEN\", {\n              channelType,\n              reliability,\n              streamId,\n              label,\n              protocol,\n            });\n\n            const maxRetransmits =\n              (channelType & 0x03) === 1 ? reliability : undefined;\n            const maxPacketLifeTime =\n              (channelType & 0x03) === 2 ? reliability : undefined;\n\n            // # register channel\n            const parameters = new RTCDataChannelParameters({\n              label,\n              ordered: (channelType & 0x80) === 0,\n              maxPacketLifeTime,\n              maxRetransmits,\n              protocol,\n              id: streamId,\n            });\n            const channel = new RTCDataChannel(this, parameters, false);\n            channel.isCreatedByRemote = true;\n            this.dataChannels[streamId] = channel;\n\n            this.onDataChannel.execute(channel);\n            channel.setReadyState(\"open\");\n          } else {\n            log(\"datachannel already opened\", \"retransmit ack\");\n          }\n\n          const channel = this.dataChannels[streamId];\n          this.dataChannelQueue.push([\n            channel,\n            WEBRTC_DCEP,\n            Buffer.from(jspack.Pack(\"!B\", [DATA_CHANNEL_ACK])),\n          ]);\n          await this.dataChannelFlush();\n        }\n        break;\n        case DATA_CHANNEL_ACK: {\n          log(\"DATA_CHANNEL_ACK\", streamId, ppId);\n          const channel = this.dataChannels[streamId];\n          if (!channel) {\n            throw new Error(\"channel not found\");\n          }\n          channel.setReadyState(\"open\");\n        }\n        break;\n      }\n    } else {\n      const channel = this.dataChannels[streamId];\n      if (channel) {\n        const msg = (() => {\n          switch (ppId) {\n            case WEBRTC_STRING:\n              return data.toString(\"utf8\");\n            case WEBRTC_STRING_EMPTY:\n              return \"\";\n            case WEBRTC_BINARY:\n              return data;\n            case WEBRTC_BINARY_EMPTY:\n              return Buffer.from([]);\n            default:\n              throw new Error();\n          }\n        })();\n\n        channel.onMessage.execute(msg);\n        channel.emit(\"message\", { data: msg });\n        if (channel.onmessage) {\n          channel.onmessage({ data: msg });\n        }\n      }\n    }\n  };\n\n  dataChannelAddNegotiated(channel: RTCDataChannel) {\n    if (channel.id == undefined) {\n      throw new Error();\n    }\n    if (this.dataChannels[channel.id]) {\n      throw new Error();\n    }\n\n    this.dataChannels[channel.id] = channel;\n\n    if (this.sctp.associationState === SCTP_STATE.ESTABLISHED) {\n      channel.setReadyState(\"open\");\n    }\n  }\n\n  dataChannelOpen(channel: RTCDataChannel) {\n    if (channel.id) {\n      if (this.dataChannels[channel.id])\n        throw new Error(\n          `Data channel with ID ${channel.id} already registered`,\n        );\n      this.dataChannels[channel.id] = channel;\n    }\n\n    let channelType = DATA_CHANNEL_RELIABLE;\n    const priority = 0;\n    let reliability = 0;\n\n    if (!channel.ordered) {\n      channelType = 0x80;\n    }\n    if (channel.maxRetransmits) {\n      channelType = 1;\n      reliability = channel.maxRetransmits;\n    } else if (channel.maxPacketLifeTime) {\n      channelType = 2;\n      reliability = channel.maxPacketLifeTime;\n    }\n\n    // 5.1.  DATA_CHANNEL_OPEN Message\n    const data = jspack.Pack(\"!BBHLHH\", [\n      DATA_CHANNEL_OPEN,\n      channelType,\n      priority,\n      reliability,\n      channel.label.length,\n      channel.protocol.length,\n    ]);\n    const send = Buffer.concat([\n      Buffer.from(data),\n      Buffer.from(channel.label, \"utf8\"),\n      Buffer.from(channel.protocol, \"utf8\"),\n    ]);\n    this.dataChannelQueue.push([channel, WEBRTC_DCEP, send]);\n    this.dataChannelFlush();\n  }\n\n  private async dataChannelFlush() {\n    // \"\"\"\n    // Try to flush buffered data to the SCTP layer.\n\n    // We wait until the association is established, as we need to know\n    // whether we are a client or a server to correctly assign an odd/even ID\n    // to the data channels.\n    // \"\"\"\n\n    if (this.sctp.associationState != SCTP_STATE.ESTABLISHED) return;\n    if (this.sctp.outboundQueue.length > 0) return;\n\n    while (this.dataChannelQueue.length > 0) {\n      const [channel, protocol, userData] = this.dataChannelQueue.shift()!;\n\n      let streamId = channel.id;\n      if (streamId === undefined) {\n        streamId = this.dataChannelId!;\n        while (Object.keys(this.dataChannels).includes(streamId.toString())) {\n          streamId += 2;\n        }\n        this.dataChannels[streamId] = channel;\n        channel.setId(streamId);\n      }\n\n      if (protocol === WEBRTC_DCEP) {\n        await this.sctp.send(streamId, protocol, userData, {\n          ordered: true,\n        });\n      } else {\n        const expiry = channel.maxPacketLifeTime\n          ? Date.now() + channel.maxPacketLifeTime / 1000\n          : undefined;\n\n        await this.sctp.send(streamId, protocol, userData, {\n          expiry,\n          maxRetransmits: channel.maxRetransmits,\n          ordered: channel.ordered,\n        });\n        channel.addBufferedAmount(-userData.length);\n      }\n    }\n  }\n\n  datachannelSend = (channel: RTCDataChannel, data: Buffer | string) => {\n    channel.addBufferedAmount(data.length);\n\n    this.dataChannelQueue.push(\n      typeof data === \"string\"\n        ? [channel, WEBRTC_STRING, Buffer.from(data)]\n        : [channel, WEBRTC_BINARY, data],\n    );\n\n    if (this.sctp.associationState !== SCTP_STATE.ESTABLISHED) {\n      log(\"sctp not established\", this.sctp.associationState);\n    }\n\n    this.dataChannelFlush();\n  };\n\n  static getCapabilities() {\n    return new RTCSctpCapabilities(65536);\n  }\n\n  setRemotePort(port: number) {\n    this.sctp.setRemotePort(port);\n  }\n\n  async start(remotePort: number) {\n    if (this.isServer) {\n      this.dataChannelId = 0;\n    } else {\n      this.dataChannelId = 1;\n    }\n    this.sctp.isServer = this.isServer;\n\n    await this.sctp.start(remotePort);\n  }\n\n  async stop() {\n    this.dtlsTransport.dataReceiver = () => {};\n    await this.sctp.stop();\n  }\n\n  dataChannelClose(channel: RTCDataChannel) {\n    if (!([\"closing\", \"closed\"] as DCState[]).includes(channel.readyState)) {\n      channel.setReadyState(\"closing\");\n\n      if (this.sctp.associationState === SCTP_STATE.ESTABLISHED) {\n        this.sctp.reconfigQueue.push(channel.id);\n        if (this.sctp.reconfigQueue.length === 1) {\n          this.sctp.transmitReconfigRequest();\n        }\n      } else {\n        this.dataChannelQueue = this.dataChannelQueue.filter(\n          (queueItem) => queueItem[0].id !== channel.id,\n        );\n        if (channel.id) {\n          delete this.dataChannels[channel.id];\n        }\n        channel.setReadyState(\"closed\");\n      }\n    }\n  }\n}\n\nexport class RTCSctpCapabilities {\n  constructor(public maxMessageSize: number) {}\n}\n\nclass BridgeDtls implements Transport {\n  constructor(private dtls: RTCDtlsTransport) {}\n  set onData(onData: (buf: Buffer) => void) {\n    this.dtls.dataReceiver = onData;\n  }\n  readonly send = (data: Buffer) => {\n    return this.dtls.sendData(data);\n  };\n  close() {}\n}\n"]}

package/lib/webrtc/src/utils.js

@@ -6,8 +6,8 @@ Object.defineProperty(exports, "__esModule", { value: true });
 exports.deepMerge = exports.MediaStreamTrackFactory = exports.createSelfSignedCertificate = exports.parseIceServers = exports.compactNtp = exports.ntpTime = exports.timestampSeconds = exports.milliTime = exports.microTime = exports.reverseDirection = exports.andDirection = exports.reverseSimulcastDirection = exports.isDtls = exports.fingerprint = void 0;
 /* eslint-disable prefer-const */
 const crypto_1 = require("crypto");
-const debug_1 = __importDefault(require("debug"));
 const dgram_1 = require("dgram");
+const debug_1 = __importDefault(require("debug"));
 const mergeWith_1 = __importDefault(require("lodash/mergeWith"));
 const perf_hooks_1 = require("perf_hooks");
 const src_1 = require("../../common/src");

package/lib/webrtc/src/utils.js.map

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function isDtls(buf: Buffer) {\n  const firstByte = buf[0];\n  return firstByte > 19 && firstByte < 64;\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\n/**\n *\n * @param signatureHash\n * @param namedCurveAlgorithm necessary when use ecdsa\n */\nexport const createSelfSignedCertificate =\n  CipherContext.createSelfSignedCertificateWithKey;\n\nexport class MediaStreamTrackFactory {\n  static async rtpSource({\n    port,\n    kind,\n  }: {\n    port?: number;\n    kind: \"audio\" | \"video\";\n  }) {\n    port ??= await randomPort();\n    const track = new MediaStreamTrack({ kind });\n\n    const udp = createSocket(\"udp4\");\n    udp.bind(port);\n    const onMessage = (msg: Buffer, rinfo: RemoteInfo) => {\n      track.writeRtp(msg);\n    };\n    udp.addListener(\"message\", onMessage);\n\n    const dispose = () => {\n      udp.removeListener(\"message\", onMessage);\n      udp.close();\n    };\n\n    return [track, port, dispose] as const;\n  }\n}\n\nexport const deepMerge = <T>(dst: T, src: T) =>\n  mergeWith(dst, src, (obj, src) => {\n    if (!(src == undefined)) {\n      return src;\n    }\n    return obj;\n  });\n"]}
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eslint-disable prefer-const */\nimport { createHash } from \"crypto\";\nimport { RemoteInfo, createSocket } from \"dgram\";\nimport debug from \"debug\";\nimport mergeWith from \"lodash/mergeWith\";\nimport { performance } from \"perf_hooks\";\n\nimport { bufferReader, bufferWriter, randomPort } from \"../../common/src\";\nimport { CipherContext } from \"../../dtls/src/context/cipher\";\nimport { Address } from \"../../ice/src\";\nimport { Direction, Directions } from \"./media/rtpTransceiver\";\nimport { MediaStreamTrack } from \"./media/track\";\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 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\n/**\n *\n * @param signatureHash\n * @param namedCurveAlgorithm necessary when use ecdsa\n */\nexport const createSelfSignedCertificate =\n  CipherContext.createSelfSignedCertificateWithKey;\n\nexport class MediaStreamTrackFactory {\n  static async rtpSource({\n    port,\n    kind,\n  }: {\n    port?: number;\n    kind: \"audio\" | \"video\";\n  }) {\n    port ??= await randomPort();\n    const track = new MediaStreamTrack({ kind });\n\n    const udp = createSocket(\"udp4\");\n    udp.bind(port);\n    const onMessage = (msg: Buffer, rinfo: RemoteInfo) => {\n      track.writeRtp(msg);\n    };\n    udp.addListener(\"message\", onMessage);\n\n    const dispose = () => {\n      udp.removeListener(\"message\", onMessage);\n      udp.close();\n    };\n\n    return [track, port, dispose] as const;\n  }\n}\n\nexport const deepMerge = <T>(dst: T, src: T) =>\n  mergeWith(dst, src, (obj, src) => {\n    if (!(src == undefined)) {\n      return src;\n    }\n    return obj;\n  });\n"]}

package/package.json

@@ -1,11 +1,8 @@
 {
   "name": "werift",
-  "version": "0.19.0",
+  "version": "0.19.1",
   "description": "WebRTC Implementation for TypeScript (Node.js)",
-  "keywords": [
-    "WebRTC",
-    "node.js"
-  ],
+  "keywords": ["WebRTC", "node.js"],
   "homepage": "https://github.com/shinyoshiaki/werift-webrtc",
   "repository": {
     "type": "git",
@@ -17,29 +14,25 @@
   },
   "exports": {
     ".": {
-      "default": "./lib/webrtc/src/index.js",
-      "types": "./lib/webrtc/src/index.d.ts"
+      "types": "./lib/webrtc/src/index.d.ts",     
+       "default": "./lib/webrtc/src/index.js"
     },
     "./nonstandard": {
-      "default": "./lib/webrtc/src/nonstandard/index.js",
-      "types": "./lib/webrtc/src/nonstandard/index.d.ts"
+      "types": "./lib/webrtc/src/nonstandard/index.d.ts",
+      "default": "./lib/webrtc/src/nonstandard/index.js"
     }
   },
   "main": "lib/webrtc/src/index.js",
   "types": "lib/webrtc/src/index.d.ts",
-  "files": [
-    "lib",
-    "src"
-  ],
+  "files": ["lib", "src"],
   "scripts": {
     "build": "npm i && rm -rf lib && npm run format && tsc -p ./tsconfig.production.json",
     "depcheck": "npx depcheck --ignores='@typescript-eslint/*,eslint-*'",
     "deploy": "npm run build && npm publish",
     "deploy:alpha": "npm run build && npm publish --tag alpha",
     "doc": "rm -rf doc && typedoc --excludePrivate --disableSources --excludeInternal --plugin typedoc-plugin-markdown --out doc ./src/index.ts",
-    "format": "npm run unused && npm run lint && prettier --write src/**/*.ts",
+    "format": "biome check --apply src tests",
     "knip": "knip",
-    "lint": "eslint ./src --fix",
     "test": "npm run type && jest --runInBand --forceExit --coverage",
     "type": "tsc --noEmit --project ./tsconfig.json",
     "unused": "organize-imports-cli src/**/*.ts",
@@ -54,7 +47,7 @@
     "@shinyoshiaki/ebml-builder": "^0.0.1",
     "aes-js": "^3.1.2",
     "binary-data": "^0.6.0",
-    "buffer-crc32": "^0.2.13",
+    "buffer-crc32": "^1.0.0",
     "date-fns": "^2.29.3",
     "debug": "^4.3.4",
     "int64-buffer": "^1.0.1",
@@ -71,7 +64,6 @@
   },
   "devDependencies": {
     "@types/aes-js": "^3.1.1",
-    "@types/buffer-crc32": "^0.2.2",
     "@types/debug": "^4.1.7",
     "@types/ip": "^1.1.0",
     "@types/jest": "^29.2.4",

package/src/dataChannel.ts

@@ -9,7 +9,7 @@ const log = debug("werift:packages/webrtc/src/dataChannel.ts");
 
 export class RTCDataChannel extends EventTarget {
   readonly stateChanged = new Event<[DCState]>();
-  readonly message = new Event<[string | Buffer]>();
+  readonly onMessage = new Event<[string | Buffer]>();
   // todo impl
   readonly error = new Event<[Error]>();
   readonly bufferedAmountLow = new Event();

package/src/media/parameters.ts

@@ -60,7 +60,7 @@ export class RTCRtpHeaderExtensionParameters {
 
 export class RTCRtcpParameters {
   cname?: string;
-  mux: boolean = false;
+  mux = false;
   ssrc?: number;
 
   constructor(props: Partial<RTCRtcpParameters> = {}) {
@@ -101,7 +101,7 @@ export interface RTCRtpReceiveParameters extends RTCRtpParameters {
   encodings: RTCRtpCodingParameters[];
 }
 
-export interface RTCRtpSendParameters extends RTCRtpParameters {}
+export type RTCRtpSendParameters = RTCRtpParameters;
 
 export class RTCRtpSimulcastParameters {
   rid!: string;

package/src/media/receiver/nack.ts

@@ -90,13 +90,10 @@ export class NackHandler {
     if (this.lostSeqNumbers.length > LOST_SIZE) {
       this._lost = Object.entries(this._lost)
         .slice(-LOST_SIZE)
-        .reduce(
-          (acc, [key, v]) => {
-            acc[key] = v;
-            return acc;
-          },
-          {} as { [seqNum: number]: number },
-        );
+        .reduce((acc, [key, v]) => {
+          acc[key] = v;
+          return acc;
+        }, {} as { [seqNum: number]: number });
     }
   }
 

package/src/media/router.ts

@@ -1,6 +1,7 @@
 import debug from "debug";
 
 import {
+  RTP_EXTENSION_URI,
   ReceiverEstimatedMaxBitrate,
   RtcpPacket,
   RtcpPayloadSpecificFeedback,
@@ -8,9 +9,8 @@ import {
   RtcpSourceDescriptionPacket,
   RtcpSrPacket,
   RtcpTransportLayerFeedback,
-  RTP_EXTENSION_URI,
-  rtpHeaderExtensionsParser,
   RtpPacket,
+  rtpHeaderExtensionsParser,
 } from "../../../rtp/src";
 import {
   RTCRtpReceiveParameters,
@@ -144,47 +144,43 @@ export class RtpRouter {
     const recipients: (RTCRtpReceiver | RTCRtpSender)[] = [];
 
     switch (packet.type) {
-      case RtcpSrPacket.type:
-        {
-          packet = packet as RtcpSrPacket;
-          recipients.push(this.ssrcTable[packet.ssrc]);
-        }
-        break;
-      case RtcpRrPacket.type:
-        {
-          packet = packet as RtcpRrPacket;
-          packet.reports.forEach((report) => {
-            recipients.push(this.ssrcTable[report.ssrc]);
-          });
-        }
-        break;
-      case RtcpSourceDescriptionPacket.type:
-        {
-          const sdes = packet as RtcpSourceDescriptionPacket;
-          // log("sdes", JSON.stringify(sdes.chunks));
-        }
-        break;
-      case RtcpTransportLayerFeedback.type:
-        {
-          const rtpfb = packet as RtcpTransportLayerFeedback;
-          if (rtpfb.feedback) {
-            recipients.push(this.ssrcTable[rtpfb.feedback.mediaSourceSsrc]);
-          }
+      case RtcpSrPacket.type: {
+        packet = packet as RtcpSrPacket;
+        recipients.push(this.ssrcTable[packet.ssrc]);
+      }
+      break;
+      case RtcpRrPacket.type: {
+        packet = packet as RtcpRrPacket;
+        packet.reports.forEach((report) => {
+          recipients.push(this.ssrcTable[report.ssrc]);
+        });
+      }
+      break;
+      case RtcpSourceDescriptionPacket.type: {
+        const sdes = packet as RtcpSourceDescriptionPacket;
+        // log("sdes", JSON.stringify(sdes.chunks));
+      }
+      break;
+      case RtcpTransportLayerFeedback.type: {
+        const rtpfb = packet as RtcpTransportLayerFeedback;
+        if (rtpfb.feedback) {
+          recipients.push(this.ssrcTable[rtpfb.feedback.mediaSourceSsrc]);
         }
-        break;
-      case RtcpPayloadSpecificFeedback.type:
-        {
-          const psfb = packet as RtcpPayloadSpecificFeedback;
-          switch (psfb.feedback.count) {
-            case ReceiverEstimatedMaxBitrate.count:
-              const remb = psfb.feedback as ReceiverEstimatedMaxBitrate;
-              recipients.push(this.ssrcTable[remb.ssrcFeedbacks[0]]);
-              break;
-            default:
-              recipients.push(this.ssrcTable[psfb.feedback.senderSsrc]);
+      }
+      break;
+      case RtcpPayloadSpecificFeedback.type: {
+        const psfb = packet as RtcpPayloadSpecificFeedback;
+        switch (psfb.feedback.count) {
+          case ReceiverEstimatedMaxBitrate.count: {
+            const remb = psfb.feedback as ReceiverEstimatedMaxBitrate;
+            recipients.push(this.ssrcTable[remb.ssrcFeedbacks[0]]);
           }
+          break;
+          default:
+            recipients.push(this.ssrcTable[psfb.feedback.senderSsrc]);
         }
-        break;
+      }
+      break;
     }
     recipients
       .filter((v) => v) // todo simulcast

package/src/media/rtpReceiver.ts

@@ -3,9 +3,11 @@ import Event from "rx.mini";
 import { setTimeout } from "timers/promises";
 import { v4 as uuid } from "uuid";
 
+import { PeerConfig, codecParametersFromString, usePLI, useTWCC } from "..";
 import { int } from "../../../common/src";
 import {
   PictureLossIndication,
+  RTP_EXTENSION_URI,
   Red,
   RedHandler,
   RtcpPacket,
@@ -13,12 +15,10 @@ import {
   RtcpReceiverInfo,
   RtcpRrPacket,
   RtcpSrPacket,
-  RTP_EXTENSION_URI,
   RtpPacket,
   TransportWideCCPayload,
   unwrapRtx,
 } from "../../../rtp/src";
-import { codecParametersFromString, PeerConfig, usePLI, useTWCC } from "..";
 import { RTCDtlsTransport } from "../transport/dtls";
 import { Kind } from "../types/domain";
 import { compactNtp, timestampSeconds } from "../utils";
@@ -236,20 +236,17 @@ export class RTCRtpReceiver {
 
   handleRtcpPacket(packet: RtcpPacket) {
     switch (packet.type) {
-      case RtcpSrPacket.type:
-        {
-          const sr = packet as RtcpSrPacket;
-          this.lastSRtimestamp[sr.ssrc] = compactNtp(
-            sr.senderInfo.ntpTimestamp,
-          );
-          this.receiveLastSRTimestamp[sr.ssrc] = timestampSeconds();
-
-          const track = this.trackBySSRC[packet.ssrc];
-          if (track) {
-            track.onReceiveRtcp.execute(packet);
-          }
+      case RtcpSrPacket.type: {
+        const sr = packet as RtcpSrPacket;
+        this.lastSRtimestamp[sr.ssrc] = compactNtp(sr.senderInfo.ntpTimestamp);
+        this.receiveLastSRTimestamp[sr.ssrc] = timestampSeconds();
+
+        const track = this.trackBySSRC[packet.ssrc];
+        if (track) {
+          track.onReceiveRtcp.execute(packet);
         }
-        break;
+      }
+      break;
     }
     this.onRtcp.execute(packet);
   }

package/src/media/rtpSender.ts

@@ -27,11 +27,13 @@ import Event from "rx.mini";
 import { setTimeout } from "timers/promises";
 import * as uuid from "uuid";
 
+import { codecParametersFromString } from "..";
 import { random16, uint16Add, uint32Add } from "../../../common/src";
 import {
   Extension,
   GenericNack,
   PictureLossIndication,
+  RTP_EXTENSION_URI,
   ReceiverEstimatedMaxBitrate,
   RedEncoder,
   RtcpPacket,
@@ -41,20 +43,18 @@ import {
   RtcpSourceDescriptionPacket,
   RtcpSrPacket,
   RtcpTransportLayerFeedback,
-  RTP_EXTENSION_URI,
   RtpHeader,
   RtpPacket,
+  SourceDescriptionChunk,
+  SourceDescriptionItem,
+  TransportWideCC,
   serializeAbsSendTime,
   serializeRepairedRtpStreamId,
   serializeSdesMid,
   serializeSdesRTPStreamID,
   serializeTransportWideCC,
-  SourceDescriptionChunk,
-  SourceDescriptionItem,
-  TransportWideCC,
   wrapRtx,
 } from "../../../rtp/src";
-import { codecParametersFromString } from "..";
 import { RTCDtlsTransport } from "../transport/dtls";
 import { Kind } from "../types/domain";
 import { compactNtp, milliTime, ntpTime, timestampSeconds } from "../utils";
@@ -104,7 +104,7 @@ export class RTCRtpSender {
   private octetCount = 0;
   private packetCount = 0;
   private rtt?: number;
-  receiverEstimatedMaxBitrate: bigint = 0n;
+  receiverEstimatedMaxBitrate = 0n;
 
   // rtp
   private sequenceNumber?: number;
@@ -410,90 +410,77 @@ export class RTCRtpSender {
   handleRtcpPacket(rtcpPacket: RtcpPacket) {
     switch (rtcpPacket.type) {
       case RtcpSrPacket.type:
-      case RtcpRrPacket.type:
-        {
-          const packet = rtcpPacket as RtcpSrPacket | RtcpRrPacket;
-          packet.reports
-            .filter((report) => report.ssrc === this.ssrc)
-            .forEach((report) => {
-              if (this.lastSRtimestamp === report.lsr && report.dlsr) {
-                if (this.lastSentSRTimestamp) {
-                  const rtt =
-                    timestampSeconds() -
-                    this.lastSentSRTimestamp -
-                    report.dlsr / 65536;
-                  if (this.rtt === undefined) {
-                    this.rtt = rtt;
-                  } else {
-                    this.rtt = RTT_ALPHA * this.rtt + (1 - RTT_ALPHA) * rtt;
-                  }
+      case RtcpRrPacket.type: {
+        const packet = rtcpPacket as RtcpSrPacket | RtcpRrPacket;
+        packet.reports
+          .filter((report) => report.ssrc === this.ssrc)
+          .forEach((report) => {
+            if (this.lastSRtimestamp === report.lsr && report.dlsr) {
+              if (this.lastSentSRTimestamp) {
+                const rtt =
+                  timestampSeconds() -
+                  this.lastSentSRTimestamp -
+                  report.dlsr / 65536;
+                if (this.rtt === undefined) {
+                  this.rtt = rtt;
+                } else {
+                  this.rtt = RTT_ALPHA * this.rtt + (1 - RTT_ALPHA) * rtt;
                 }
               }
-            });
-        }
-        break;
-      case RtcpTransportLayerFeedback.type:
-        {
-          const packet = rtcpPacket as RtcpTransportLayerFeedback;
-          switch (packet.feedback.count) {
-            case TransportWideCC.count:
-              {
-                const feedback = packet.feedback as TransportWideCC;
-                this.senderBWE.receiveTWCC(feedback);
+            }
+          });
+      }
+      break;
+      case RtcpTransportLayerFeedback.type: {
+        const packet = rtcpPacket as RtcpTransportLayerFeedback;
+        switch (packet.feedback.count) {
+          case TransportWideCC.count: {
+            const feedback = packet.feedback as TransportWideCC;
+            this.senderBWE.receiveTWCC(feedback);
+          }
+          break;
+          case GenericNack.count: {
+            const feedback = packet.feedback as GenericNack;
+            feedback.lost.forEach(async (seqNum) => {
+              let packet: RtpPacket | undefined =
+                this.rtpCache[seqNum % RTP_HISTORY_SIZE];
+              if (packet && packet.header.sequenceNumber !== seqNum) {
+                packet = undefined;
               }
-              break;
-            case GenericNack.count:
-              {
-                const feedback = packet.feedback as GenericNack;
-                feedback.lost.forEach(async (seqNum) => {
-                  let packet: RtpPacket | undefined =
-                    this.rtpCache[seqNum % RTP_HISTORY_SIZE];
-                  if (packet && packet.header.sequenceNumber !== seqNum) {
-                    packet = undefined;
-                  }
-                  if (packet) {
-                    if (this.rtxPayloadType != undefined) {
-                      packet = wrapRtx(
-                        packet,
-                        this.rtxPayloadType,
-                        this.rtxSequenceNumber,
-                        this.rtxSsrc,
-                      );
-                      this.rtxSequenceNumber = uint16Add(
-                        this.rtxSequenceNumber,
-                        1,
-                      );
-                    }
-                    await this.dtlsTransport.sendRtp(
-                      packet.payload,
-                      packet.header,
-                    );
-                  }
-                });
-                this.onGenericNack.execute(feedback);
+              if (packet) {
+                if (this.rtxPayloadType != undefined) {
+                  packet = wrapRtx(
+                    packet,
+                    this.rtxPayloadType,
+                    this.rtxSequenceNumber,
+                    this.rtxSsrc,
+                  );
+                  this.rtxSequenceNumber = uint16Add(this.rtxSequenceNumber, 1);
+                }
+                await this.dtlsTransport.sendRtp(packet.payload, packet.header);
               }
-              break;
+            });
+            this.onGenericNack.execute(feedback);
           }
+          break;
         }
-        break;
-      case RtcpPayloadSpecificFeedback.type:
-        {
-          const packet = rtcpPacket as RtcpPayloadSpecificFeedback;
-          switch (packet.feedback.count) {
-            case ReceiverEstimatedMaxBitrate.count:
-              {
-                const feedback = packet.feedback as ReceiverEstimatedMaxBitrate;
-                this.receiverEstimatedMaxBitrate = feedback.bitrate;
-              }
-              break;
-            case PictureLossIndication.count:
-              {
-                this.onPictureLossIndication.execute();
-              }
-              break;
+      }
+      break;
+      case RtcpPayloadSpecificFeedback.type: {
+        const packet = rtcpPacket as RtcpPayloadSpecificFeedback;
+        switch (packet.feedback.count) {
+          case ReceiverEstimatedMaxBitrate.count: {
+            const feedback = packet.feedback as ReceiverEstimatedMaxBitrate;
+            this.receiverEstimatedMaxBitrate = feedback.bitrate;
           }
+          break;
+          case PictureLossIndication.count: {
+            this.onPictureLossIndication.execute();
+          }
+          break;
         }
-        break;
+      }
+      break;
     }
     this.onRtcp.execute(rtcpPacket);
   }

package/src/nonstandard/recorder/writer/index.ts

@@ -1,5 +1,5 @@
-import { MediaStreamTrack } from "../../..";
 import { MediaRecorderOptions } from "..";
+import { MediaStreamTrack } from "../../..";
 
 export abstract class MediaWriter {
   constructor(