werift 0.13.5 → 0.14.1-debug

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

@@ -11,10 +11,12 @@ const promises_1 = require("timers/promises");
 const uuid_1 = require("uuid");
 const src_1 = require("../../../common/src");
 const src_2 = require("../../../rtp/src");
+const __1 = require("..");
 const utils_1 = require("../utils");
 const rtpExtension_1 = require("./extension/rtpExtension");
 const nack_1 = require("./receiver/nack");
 const receiverTwcc_1 = require("./receiver/receiverTwcc");
+const red_1 = require("./receiver/red");
 const statistics_1 = require("./receiver/statistics");
 const log = (0, debug_1.debug)("werift:packages/webrtc/src/media/rtpReceiver.ts");
 class RTCRtpReceiver {
@@ -25,6 +27,7 @@ class RTCRtpReceiver {
         this.codecs = {};
         this.ssrcByRtx = {};
         this.nack = new nack_1.Nack(this);
+        this.redHandler = new red_1.RedHandler();
         this.type = "receiver";
         this.uuid = (0, uuid_1.v4)();
         this.tracks = [];
@@ -196,17 +199,31 @@ class RTCRtpReceiver {
         }
         if (codec.name.toLowerCase() === "rtx") {
             const originalSsrc = this.ssrcByRtx[packet.header.ssrc];
-            const rtxCodec = this.codecs[codec.parameters["apt"]];
+            const codecParams = (0, __1.codecParametersFromString)(codec.parameters ?? "");
+            const rtxCodec = this.codecs[codecParams["apt"]];
             if (packet.payload.length < 2)
                 return;
             packet = unwrapRtx(packet, rtxCodec.payloadType, originalSsrc);
             track = this.trackBySSRC[originalSsrc];
         }
+        let red;
+        if (codec.name.toLowerCase() === "red") {
+            red = src_2.Red.deSerialize(packet.payload);
+        }
         // todo fix select use or not use nack
-        if (track?.kind === "video")
+        if (track?.kind === "video") {
             this.nack.addPacket(packet);
+        }
         if (track) {
-            track.onReceiveRtp.execute(packet.clone());
+            if (red) {
+                const payloads = this.redHandler.push(red, packet);
+                for (const packet of payloads) {
+                    track.onReceiveRtp.execute(packet.clone());
+                }
+            }
+            else {
+                track.onReceiveRtp.execute(packet.clone());
+            }
         }
         this.runRtcp();
     }

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  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  RtpHeader,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport { RTCRtpCodecParameters, RTCRtpReceiveParameters } from \"./parameters\";\nimport { Nack } 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 readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new Nack(this);\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 senderReport\n  readonly lsr: { [ssrc: number]: number } = {};\n  readonly lsrTime: { [ssrc: number]: number } = {};\n  readonly onPacketLost = this.nack.onPacketLost;\n  readonly onRtcp = new Event<[RtcpPacket]>();\n\n  sdesMid?: string;\n  latestRid?: string;\n  latestRepairedRid?: string;\n\n  receiverTWCC?: ReceiverTWCC;\n  supportTWCC = false;\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    public kind: Kind,\n    public dtlsTransport: RTCDtlsTransport,\n    public rtcpSsrc: number\n  ) {}\n\n  // todo fix\n  get track() {\n    return this.tracks[0];\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    this.supportTWCC = !!Object.values(this.codecs).find((codec) =>\n      codec.rtcpFeedback.find((v) => v.type === \"transport-cc\")\n    );\n    log(\"twcc support\", this.supportTWCC);\n\n    if (this.supportTWCC && mediaSourceSsrc) {\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) return t.rid === track.rid;\n      if (t.ssrc) return t.ssrc === track.ssrc;\n    });\n    if (exist) return false;\n    this.tracks.push(track);\n    if (track.ssrc) this.trackBySSRC[track.ssrc] = track;\n    if (track.rid) this.trackByRID[track.rid] = track;\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 lsr = 0,\n              dlsr = 0;\n            if (this.lsr[ssrc]) {\n              lsr = this.lsr[ssrc];\n              const delay = Date.now() / 1000 - this.lsrTime[ssrc];\n              if (delay > 0 && delay < 65536) {\n                dlsr = int(delay * 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,\n              dlsr,\n            });\n          }\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\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    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.lsr[sr.ssrc] = compactNtp(sr.senderInfo.ntpTimestamp);\n          this.lsrTime[sr.ssrc] = Date.now() / 1000;\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) return;\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.supportTWCC) {\n      this.setupTWCC(packet.header.ssrc);\n    }\n\n    if (codec.name.toLowerCase() === \"rtx\") {\n      const originalSsrc = this.ssrcByRtx[packet.header.ssrc];\n      const rtxCodec = this.codecs[codec.parameters[\"apt\"]];\n      if (packet.payload.length < 2) return;\n\n      packet = unwrapRtx(packet, rtxCodec.payloadType, originalSsrc);\n      track = this.trackBySSRC[originalSsrc];\n    }\n\n    // todo fix select use or not use nack\n    if (track?.kind === \"video\") this.nack.addPacket(packet);\n\n    if (track) {\n      track.onReceiveRtp.execute(packet.clone());\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.slice(0, 2))[0],\n      timestamp: rtx.header.timestamp,\n      ssrc,\n    }),\n    rtx.payload.slice(2)\n  );\n  return packet;\n}\n"]}
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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 readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new Nack(this);\n  private readonly redHandler = 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 senderReport\n  readonly lsr: { [ssrc: number]: number } = {};\n  readonly lsrTime: { [ssrc: number]: number } = {};\n  readonly onPacketLost = this.nack.onPacketLost;\n  readonly onRtcp = new Event<[RtcpPacket]>();\n\n  sdesMid?: string;\n  latestRid?: string;\n  latestRepairedRid?: string;\n\n  receiverTWCC?: ReceiverTWCC;\n  supportTWCC = false;\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    public kind: Kind,\n    public dtlsTransport: RTCDtlsTransport,\n    public rtcpSsrc: number\n  ) {}\n\n  // todo fix\n  get track() {\n    return this.tracks[0];\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    this.supportTWCC = !!Object.values(this.codecs).find((codec) =>\n      codec.rtcpFeedback.find((v) => v.type === \"transport-cc\")\n    );\n    log(\"twcc support\", this.supportTWCC);\n\n    if (this.supportTWCC && mediaSourceSsrc) {\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) return t.rid === track.rid;\n      if (t.ssrc) return t.ssrc === track.ssrc;\n    });\n    if (exist) return false;\n    this.tracks.push(track);\n    if (track.ssrc) this.trackBySSRC[track.ssrc] = track;\n    if (track.rid) this.trackByRID[track.rid] = track;\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 lsr = 0,\n              dlsr = 0;\n            if (this.lsr[ssrc]) {\n              lsr = this.lsr[ssrc];\n              const delay = Date.now() / 1000 - this.lsrTime[ssrc];\n              if (delay > 0 && delay < 65536) {\n                dlsr = int(delay * 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,\n              dlsr,\n            });\n          }\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\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    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.lsr[sr.ssrc] = compactNtp(sr.senderInfo.ntpTimestamp);\n          this.lsrTime[sr.ssrc] = Date.now() / 1000;\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) return;\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.supportTWCC) {\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    }\n\n    // todo fix select use or not use nack\n    if (track?.kind === \"video\") {\n      this.nack.addPacket(packet);\n    }\n\n    if (track) {\n      if (red) {\n        const payloads = this.redHandler.push(red, packet);\n        for (const packet of payloads) {\n          track.onReceiveRtp.execute(packet.clone());\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.slice(0, 2))[0],\n      timestamp: rtx.header.timestamp,\n      ssrc,\n    }),\n    rtx.payload.slice(2)\n  );\n  return packet;\n}\n"]}

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

@@ -1,9 +1,9 @@
 /// <reference types="node" />
 import Event from "rx.mini";
-import { GenericNack, RtcpPacket, RtpHeader, RtpPacket } from "../../../rtp/src";
+import { GenericNack, Red, RtcpPacket, RtpHeader, RtpPacket } from "../../../rtp/src";
 import { RTCDtlsTransport } from "../transport/dtls";
 import { Kind } from "../types/domain";
-import { RTCRtpSendParameters } from "./parameters";
+import { RTCRtpCodecParameters, RTCRtpSendParameters } from "./parameters";
 import { SenderBandwidthEstimator } from "./sender/senderBWE";
 import { MediaStreamTrack } from "./track";
 export declare class RTCRtpSender {
@@ -26,6 +26,8 @@ export declare class RTCRtpSender {
     private repairedRtpStreamId?;
     private rtxPayloadType?;
     private rtxSequenceNumber;
+    redRedundantPayloadType?: number;
+    redDistance: number;
     private headerExtensions;
     private disposeTrack?;
     private lsr?;
@@ -41,7 +43,7 @@ export declare class RTCRtpSender {
     private timestampOffset;
     private seqOffset;
     private rtpCache;
-    private codec?;
+    codec?: RTCRtpCodecParameters;
     track?: MediaStreamTrack;
     stopped: boolean;
     rtcpRunning: boolean;
@@ -57,3 +59,4 @@ export declare class RTCRtpSender {
     handleRtcpPacket(rtcpPacket: RtcpPacket): void;
 }
 export declare function wrapRtx(packet: RtpPacket, payloadType: number, sequenceNumber: number, ssrc: number): RtpPacket;
+export declare function buildRedPacket(redundantPackets: RtpPacket[], blockPT: number, presentPacket: RtpPacket): Red;

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

@@ -22,7 +22,7 @@ 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.buildRedPacket = exports.wrapRtx = exports.RTCRtpSender = void 0;
 const crypto_1 = require("crypto");
 const debug_1 = __importDefault(require("debug"));
 const jspack_1 = require("jspack");
@@ -31,6 +31,7 @@ const promises_1 = require("timers/promises");
 const uuid = __importStar(require("uuid"));
 const src_1 = require("../../../common/src");
 const src_2 = require("../../../rtp/src");
+const __1 = require("..");
 const utils_1 = require("../utils");
 const rtpExtension_1 = require("./extension/rtpExtension");
 const senderBWE_1 = require("./sender/senderBWE");
@@ -56,6 +57,7 @@ class RTCRtpSender {
         this.onGenericNack = new rx_mini_1.default();
         this.senderBWE = new senderBWE_1.SenderBandwidthEstimator();
         this.rtxSequenceNumber = (0, src_1.random16)();
+        this.redDistance = 2;
         this.headerExtensions = [];
         this.lsrTime = Date.now() / 1000;
         this.ntpTimestamp = 0n;
@@ -87,17 +89,20 @@ class RTCRtpSender {
         this.headerExtensions = params.headerExtensions;
         this.rtpStreamId = params.rtpStreamId;
         this.repairedRtpStreamId = params.repairedRtpStreamId;
-        for (const codec of params.codecs) {
-            if (codec.name.toLowerCase() === "rtx" &&
-                codec.parameters["apt"] === params.codecs[0].payloadType) {
-                this.rtxPayloadType = codec.payloadType;
-                break;
-            }
-        }
         this.codec = params.codecs[0];
         if (this.track) {
             this.track.codec = this.codec;
         }
+        params.codecs.forEach((codec) => {
+            const codecParams = (0, __1.codecParametersFromString)(codec.parameters ?? "");
+            if (codec.name.toLowerCase() === "rtx" &&
+                codecParams["apt"] === this.codec?.payloadType) {
+                this.rtxPayloadType = codec.payloadType;
+            }
+            if (codec.name.toLowerCase() === "red") {
+                this.redRedundantPayloadType = Number((codec.parameters ?? "").split("/")[0]);
+            }
+        });
     }
     registerTrack(track) {
         if (track.stopped)
@@ -188,7 +193,7 @@ class RTCRtpSender {
             this.seqOffset = (0, src_1.uint16Add)(this.sequenceNumber, -sequenceNumber);
         }
         if (this.timestamp != undefined) {
-            this.timestampOffset = Number((0, src_1.uint32Add)(BigInt(this.timestamp), BigInt(-timestamp)));
+            this.timestampOffset = (0, src_1.uint32Add)(this.timestamp, -timestamp);
         }
         this.rtpCache = [];
         log("replaceRTP", this.sequenceNumber, sequenceNumber, this.seqOffset);
@@ -201,7 +206,7 @@ class RTCRtpSender {
         const header = rtp.header;
         header.ssrc = this.ssrc;
         header.payloadType = this.codec.payloadType;
-        header.timestamp = Number((0, src_1.uint32Add)(BigInt(header.timestamp), BigInt(this.timestampOffset)));
+        header.timestamp = (0, src_1.uint32Add)(header.timestamp, this.timestampOffset);
         header.sequenceNumber = (0, src_1.uint16Add)(header.sequenceNumber, this.seqOffset);
         this.timestamp = header.timestamp;
         this.sequenceNumber = header.sequenceNumber;
@@ -243,11 +248,22 @@ class RTCRtpSender {
         this.ntpTimestamp = (0, utils_1.ntpTime)();
         this.rtpTimestamp = rtp.header.timestamp;
         this.octetCount += rtp.payload.length;
-        this.packetCount = Number((0, src_1.uint32Add)(BigInt(this.packetCount), 1n));
+        this.packetCount = (0, src_1.uint32Add)(this.packetCount, 1);
         rtp.header = header;
         this.rtpCache.push(rtp);
         this.rtpCache = this.rtpCache.slice(-RTP_HISTORY_SIZE);
-        const size = this.dtlsTransport.sendRtp(rtp.payload, header);
+        let rtpPayload = rtp.payload;
+        if (this.redRedundantPayloadType) {
+            const redundantPackets = [...Array(this.redDistance).keys()]
+                .map((i) => {
+                return this.rtpCache.find((c) => c.header.sequenceNumber ===
+                    header.sequenceNumber - (this.redDistance - i));
+            })
+                .filter((p) => typeof p !== "undefined");
+            const red = buildRedPacket(redundantPackets, this.redRedundantPayloadType, rtp);
+            rtpPayload = red.serialize();
+        }
+        const size = this.dtlsTransport.sendRtp(rtpPayload, header);
         this.runRtcp();
         const sentInfo = {
             wideSeq: this.dtlsTransport.transportSequenceNumber,
@@ -346,4 +362,20 @@ function wrapRtx(packet, payloadType, sequenceNumber, ssrc) {
     return rtx;
 }
 exports.wrapRtx = wrapRtx;
+function buildRedPacket(redundantPackets, blockPT, presentPacket) {
+    const red = new src_2.Red();
+    redundantPackets.forEach((redundant) => {
+        red.blocks.push({
+            block: redundant.payload,
+            blockPT,
+            timestampOffset: (0, src_1.uint32Add)(presentPacket.header.timestamp, -redundant.header.timestamp),
+        });
+    });
+    red.blocks.push({
+        block: presentPacket.payload,
+        blockPT,
+    });
+    return red;
+}
+exports.buildRedPacket = buildRedPacket;
 //# sourceMappingURL=rtpSender.js.map

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

@@ -1 +1 @@
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{ 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  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 { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, milliTime, ntpTime } 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 =\n    typeof this.trackOrKind === \"string\"\n      ? this.trackOrKind\n      : this.trackOrKind.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  private headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n  private disposeTrack?: () => void;\n\n  // # stats\n  private lsr?: number;\n  private lsrTime: number = Date.now() / 1000;\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  private codec?: RTCRtpCodecParameters;\n\n  track?: MediaStreamTrack;\n  stopped = false;\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n\n  constructor(\n    public trackOrKind: Kind | MediaStreamTrack,\n    public dtlsTransport: RTCDtlsTransport\n  ) {\n    dtlsTransport.onStateChange.subscribe((state) => {\n      if (state === \"connected\") {\n        this.onReady.execute();\n      }\n    });\n    if (trackOrKind instanceof MediaStreamTrack) {\n      if (trackOrKind.streamId) {\n        this.streamId = trackOrKind.streamId;\n      }\n      this.registerTrack(trackOrKind);\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    for (const codec of params.codecs) {\n      if (\n        codec.name.toLowerCase() === \"rtx\" &&\n        codec.parameters[\"apt\"] === params.codecs[0].payloadType\n      ) {\n        this.rtxPayloadType = codec.payloadType;\n        break;\n      }\n    }\n\n    this.codec = params.codecs[0];\n    if (this.track) {\n      this.track.codec = this.codec;\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((rtp) => {\n      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\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.lsr = compactNtp(this.ntpTimestamp);\n        this.lsrTime = Date.now() / 1000;\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    sequenceNumber,\n    timestamp,\n  }: Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">) {\n    if (this.sequenceNumber != undefined) {\n      this.seqOffset = uint16Add(this.sequenceNumber, -sequenceNumber);\n    }\n    if (this.timestamp != undefined) {\n      this.timestampOffset = Number(\n        uint32Add(BigInt(this.timestamp), BigInt(-timestamp))\n      );\n    }\n    this.rtpCache = [];\n    log(\"replaceRTP\", this.sequenceNumber, sequenceNumber, this.seqOffset);\n  }\n\n  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 = rtp.header;\n    header.ssrc = this.ssrc;\n    header.payloadType = this.codec.payloadType;\n    header.timestamp = Number(\n      uint32Add(BigInt(header.timestamp), BigInt(this.timestampOffset))\n    );\n    header.sequenceNumber = uint16Add(header.sequenceNumber, this.seqOffset);\n    this.timestamp = header.timestamp;\n    this.sequenceNumber = header.sequenceNumber;\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 = rtp.header.timestamp;\n    this.octetCount += rtp.payload.length;\n    this.packetCount = Number(uint32Add(BigInt(this.packetCount), 1n));\n\n    rtp.header = header;\n\n    this.rtpCache.push(rtp);\n    this.rtpCache = this.rtpCache.slice(-RTP_HISTORY_SIZE);\n\n    const size = this.dtlsTransport.sendRtp(rtp.payload, header);\n\n    this.runRtcp();\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.lsr === report.lsr && report.dlsr) {\n                const rtt =\n                  Date.now() / 1000 - this.lsrTime - 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        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((seqNum) => {\n                  let packet = this.rtpCache.find(\n                    (rtp) => rtp.header.sequenceNumber === seqNum\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                    this.dtlsTransport.sendRtp(packet.payload, packet.header);\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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{ 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  Red,\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 } 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 =\n    typeof this.trackOrKind === \"string\"\n      ? this.trackOrKind\n      : this.trackOrKind.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  redDistance = 2;\n  private headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n  private disposeTrack?: () => void;\n\n  // # stats\n  private lsr?: number;\n  private lsrTime: number = Date.now() / 1000;\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\n  track?: MediaStreamTrack;\n  stopped = false;\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n\n  constructor(\n    public trackOrKind: Kind | MediaStreamTrack,\n    public dtlsTransport: RTCDtlsTransport\n  ) {\n    dtlsTransport.onStateChange.subscribe((state) => {\n      if (state === \"connected\") {\n        this.onReady.execute();\n      }\n    });\n    if (trackOrKind instanceof MediaStreamTrack) {\n      if (trackOrKind.streamId) {\n        this.streamId = trackOrKind.streamId;\n      }\n      this.registerTrack(trackOrKind);\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((rtp) => {\n      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\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.lsr = compactNtp(this.ntpTimestamp);\n        this.lsrTime = Date.now() / 1000;\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    sequenceNumber,\n    timestamp,\n  }: Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">) {\n    if (this.sequenceNumber != undefined) {\n      this.seqOffset = uint16Add(this.sequenceNumber, -sequenceNumber);\n    }\n    if (this.timestamp != undefined) {\n      this.timestampOffset = uint32Add(this.timestamp, -timestamp);\n    }\n    this.rtpCache = [];\n    log(\"replaceRTP\", this.sequenceNumber, sequenceNumber, this.seqOffset);\n  }\n\n  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 = rtp.header;\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    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 = rtp.header.timestamp;\n    this.octetCount += rtp.payload.length;\n    this.packetCount = uint32Add(this.packetCount, 1);\n\n    rtp.header = header;\n\n    this.rtpCache.push(rtp);\n    this.rtpCache = this.rtpCache.slice(-RTP_HISTORY_SIZE);\n\n    let rtpPayload = rtp.payload;\n\n    if (this.redRedundantPayloadType) {\n      const redundantPackets = [...Array(this.redDistance).keys()]\n        .map((i) => {\n          return this.rtpCache.find(\n            (c) =>\n              c.header.sequenceNumber ===\n              header.sequenceNumber - (this.redDistance - i)\n          );\n        })\n        .filter((p): p is NonNullable<typeof p> => typeof p !== \"undefined\");\n      const red = buildRedPacket(\n        redundantPackets,\n        this.redRedundantPayloadType,\n        rtp\n      );\n      rtpPayload = red.serialize();\n    }\n\n    const size = this.dtlsTransport.sendRtp(rtpPayload, header);\n\n    this.runRtcp();\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.lsr === report.lsr && report.dlsr) {\n                const rtt =\n                  Date.now() / 1000 - this.lsrTime - 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        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((seqNum) => {\n                  let packet = this.rtpCache.find(\n                    (rtp) => rtp.header.sequenceNumber === seqNum\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                    this.dtlsTransport.sendRtp(packet.payload, packet.header);\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\nexport function buildRedPacket(\n  redundantPackets: RtpPacket[],\n  blockPT: number,\n  presentPacket: RtpPacket\n) {\n  const red = new Red();\n  redundantPackets.forEach((redundant) => {\n    red.blocks.push({\n      block: redundant.payload,\n      blockPT,\n      timestampOffset: uint32Add(\n        presentPacket.header.timestamp,\n        -redundant.header.timestamp\n      ),\n    });\n  });\n\n  red.blocks.push({\n    block: presentPacket.payload,\n    blockPT,\n  });\n  return red;\n}\n"]}

package/lib/webrtc/src/nonstandard/recorder/writer/webm.js

@@ -1,11 +1,31 @@
 "use strict";
+var __createBinding = (this && this.__createBinding) || (Object.create ? (function(o, m, k, k2) {
+    if (k2 === undefined) k2 = k;
+    Object.defineProperty(o, k2, { enumerable: true, get: function() { return m[k]; } });
+}) : (function(o, m, k, k2) {
+    if (k2 === undefined) k2 = k;
+    o[k2] = m[k];
+}));
+var __setModuleDefault = (this && this.__setModuleDefault) || (Object.create ? (function(o, v) {
+    Object.defineProperty(o, "default", { enumerable: true, value: v });
+}) : function(o, v) {
+    o["default"] = v;
+});
+var __importStar = (this && this.__importStar) || function (mod) {
+    if (mod && mod.__esModule) return mod;
+    var result = {};
+    if (mod != null) for (var k in mod) if (k !== "default" && Object.prototype.hasOwnProperty.call(mod, k)) __createBinding(result, mod, k);
+    __setModuleDefault(result, mod);
+    return result;
+};
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.WebmFactory = void 0;
+const fs = __importStar(require("fs/promises"));
 const __1 = require("../../..");
 const _1 = require(".");
 class WebmFactory extends _1.MediaWriter {
     start(tracks) {
-        this.webm = new __1.WebmOutput("./test.webm", tracks.map((track, i) => {
+        this.webm = new __1.WebmOutput(fs, "./test.webm", tracks.map((track, i) => {
             const trackNumber = i + 1;
             const payloadType = track.codec.payloadType;
             if (track.kind === "video") {

package/lib/webrtc/src/nonstandard/recorder/writer/webm.js.map

@@ -1 +1 @@
-{"version":3,"file":"webm.js","sourceRoot":"","sources":["../../../../../../src/nonstandard/recorder/writer/webm.ts"],"names":[],"mappings":";;;AACA,gCAKkB;AAClB,wBAAgC;AAEhC,MAAa,WAAY,SAAQ,cAAW;IAG1C,KAAK,CAAC,MAA0B;QAC9B,IAAI,CAAC,IAAI,GAAG,IAAI,cAAU,CACxB,aAAa,EACb,MAAM,CAAC,GAAG,CAAC,CAAC,KAAK,EAAE,CAAC,EAAE,EAAE;YACtB,MAAM,WAAW,GAAG,CAAC,GAAG,CAAC,CAAC;YAC1B,MAAM,WAAW,GAAG,KAAK,CAAC,KAAM,CAAC,WAAW,CAAC;YAE7C,IAAI,KAAK,CAAC,IAAI,KAAK,OAAO,EAAE;gBAC1B,MAAM,KAAK,GAAG,CAAC,GAAmB,EAAE;oBAClC,QAAQ,KAAK,CAAC,KAAK,EAAE,IAAI,CAAC,WAAW,EAAyB,EAAE;wBAC9D,KAAK,KAAK;4BACR,OAAO,KAAK,CAAC;wBACf,KAAK,KAAK;4BACR,OAAO,KAAK,CAAC;wBACf,KAAK,MAAM;4BACT,OAAO,eAAe,CAAC;wBACzB,KAAK,MAAM;4BACT,OAAO,KAAK,CAAC;wBACf;4BACE,MAAM,IAAI,KAAK,EAAE,CAAC;qBACrB;gBACH,CAAC,CAAC,EAAE,CAAC;gBACL,OAAO;oBACL,IAAI,EAAE,OAAO;oBACb,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,WAAW;oBACX,KAAK;oBACL,KAAK,EAAE,IAAI,CAAC,OAAO,CAAC,KAAK;oBACzB,MAAM,EAAE,IAAI,CAAC,OAAO,CAAC,MAAM;iBAC5B,CAAC;aACH;iBAAM;gBACL,OAAO;oBACL,IAAI,EAAE,OAAO;oBACb,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,WAAW;oBACX,KAAK,EAAE,MAAM;iBACd,CAAC;aACH;QACH,CAAC,CAAC,CACH,CAAC;QAEF,MAAM,CAAC,OAAO,CAAC,CAAC,KAAK,EAAE,EAAE;YACvB,MAAM,aAAa,GACjB,KAAK,CAAC,IAAI,KAAK,OAAO;gBACpB,CAAC,CAAC,IAAI,iBAAa,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,CAAC,CAAC,MAAM,CAAC,CAAC,IAAI,CAAC,IAAI,CAAC,IAAK,CAAC;gBACvD,CAAC,CAAC,IAAI,iBAAa,CAAC,GAAG,EAAE,CAAC,IAAI,CAAC,CAAC,IAAI,CAAC,IAAI,CAAC,IAAK,CAAC,CAAC;YACrD,IAAI,gBAAY,CAAC;gBACf,SAAS,EAAE,KAAK,CAAC,YAAY;gBAC7B,UAAU,EAAE,KAAK,CAAC,aAAa;aAChC,CAAC,CAAC,IAAI,CAAC,aAAa,CAAC,CAAC;QACzB,CAAC,CAAC,CAAC;IACL,CAAC;IAED,KAAK,CAAC,IAAI;QACR,MAAM,IAAI,CAAC,IAAK,CAAC,IAAI,EAAE,CAAC;IAC1B,CAAC;CACF;AA7DD,kCA6DC;AAED,MAAM,oBAAoB,GAAG,CAAC,MAAM,EAAE,KAAK,EAAE,KAAK,EAAE,MAAM,CAAU,CAAC","sourcesContent":["import { SupportedCodec } from \"../../../../../rtp/src/container/webm\";\nimport {\n  JitterBuffer,\n  MediaStreamTrack,\n  SampleBuilder,\n  WebmOutput,\n} from \"../../..\";\nimport { MediaWriter } from \".\";\n\nexport class WebmFactory extends MediaWriter {\n  webm?: WebmOutput;\n\n  start(tracks: MediaStreamTrack[]) {\n    this.webm = new WebmOutput(\n      \"./test.webm\",\n      tracks.map((track, i) => {\n        const trackNumber = i + 1;\n        const payloadType = track.codec!.payloadType;\n\n        if (track.kind === \"video\") {\n          const codec = ((): SupportedCodec => {\n            switch (track.codec?.name.toLowerCase() as SupportedVideoCodec) {\n              case \"vp8\":\n                return \"VP8\";\n              case \"vp9\":\n                return \"VP9\";\n              case \"h264\":\n                return \"MPEG4/ISO/AVC\";\n              case \"av1x\":\n                return \"AV1\";\n              default:\n                throw new Error();\n            }\n          })();\n          return {\n            kind: \"video\",\n            clockRate: 90000,\n            payloadType,\n            trackNumber,\n            codec,\n            width: this.options.width,\n            height: this.options.height,\n          };\n        } else {\n          return {\n            kind: \"audio\",\n            clockRate: 48000,\n            payloadType,\n            trackNumber,\n            codec: \"OPUS\",\n          };\n        }\n      })\n    );\n\n    tracks.forEach((track) => {\n      const sampleBuilder =\n        track.kind === \"video\"\n          ? new SampleBuilder((h) => !!h.marker).pipe(this.webm!)\n          : new SampleBuilder(() => true).pipe(this.webm!);\n      new JitterBuffer({\n        rtpStream: track.onReceiveRtp,\n        rtcpStream: track.onReceiveRtcp,\n      }).pipe(sampleBuilder);\n    });\n  }\n\n  async stop() {\n    await this.webm!.stop();\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = typeof supportedVideoCodecs[number];\n"]}
+{"version":3,"file":"webm.js","sourceRoot":"","sources":["../../../../../../src/nonstandard/recorder/writer/webm.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;;;;;;;AAAA,gDAAkC;AAGlC,gCAKkB;AAClB,wBAAgC;AAEhC,MAAa,WAAY,SAAQ,cAAW;IAG1C,KAAK,CAAC,MAA0B;QAC9B,IAAI,CAAC,IAAI,GAAG,IAAI,cAAU,CACxB,EAAE,EACF,aAAa,EACb,MAAM,CAAC,GAAG,CAAC,CAAC,KAAK,EAAE,CAAC,EAAE,EAAE;YACtB,MAAM,WAAW,GAAG,CAAC,GAAG,CAAC,CAAC;YAC1B,MAAM,WAAW,GAAG,KAAK,CAAC,KAAM,CAAC,WAAW,CAAC;YAE7C,IAAI,KAAK,CAAC,IAAI,KAAK,OAAO,EAAE;gBAC1B,MAAM,KAAK,GAAG,CAAC,GAAmB,EAAE;oBAClC,QAAQ,KAAK,CAAC,KAAK,EAAE,IAAI,CAAC,WAAW,EAAyB,EAAE;wBAC9D,KAAK,KAAK;4BACR,OAAO,KAAK,CAAC;wBACf,KAAK,KAAK;4BACR,OAAO,KAAK,CAAC;wBACf,KAAK,MAAM;4BACT,OAAO,eAAe,CAAC;wBACzB,KAAK,MAAM;4BACT,OAAO,KAAK,CAAC;wBACf;4BACE,MAAM,IAAI,KAAK,EAAE,CAAC;qBACrB;gBACH,CAAC,CAAC,EAAE,CAAC;gBACL,OAAO;oBACL,IAAI,EAAE,OAAO;oBACb,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,WAAW;oBACX,KAAK;oBACL,KAAK,EAAE,IAAI,CAAC,OAAO,CAAC,KAAK;oBACzB,MAAM,EAAE,IAAI,CAAC,OAAO,CAAC,MAAM;iBAC5B,CAAC;aACH;iBAAM;gBACL,OAAO;oBACL,IAAI,EAAE,OAAO;oBACb,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,WAAW;oBACX,KAAK,EAAE,MAAM;iBACd,CAAC;aACH;QACH,CAAC,CAAC,CACH,CAAC;QAEF,MAAM,CAAC,OAAO,CAAC,CAAC,KAAK,EAAE,EAAE;YACvB,MAAM,aAAa,GACjB,KAAK,CAAC,IAAI,KAAK,OAAO;gBACpB,CAAC,CAAC,IAAI,iBAAa,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,CAAC,CAAC,MAAM,CAAC,CAAC,IAAI,CAAC,IAAI,CAAC,IAAK,CAAC;gBACvD,CAAC,CAAC,IAAI,iBAAa,CAAC,GAAG,EAAE,CAAC,IAAI,CAAC,CAAC,IAAI,CAAC,IAAI,CAAC,IAAK,CAAC,CAAC;YACrD,IAAI,gBAAY,CAAC;gBACf,SAAS,EAAE,KAAK,CAAC,YAAY;gBAC7B,UAAU,EAAE,KAAK,CAAC,aAAa;aAChC,CAAC,CAAC,IAAI,CAAC,aAAa,CAAC,CAAC;QACzB,CAAC,CAAC,CAAC;IACL,CAAC;IAED,KAAK,CAAC,IAAI;QACR,MAAM,IAAI,CAAC,IAAK,CAAC,IAAI,EAAE,CAAC;IAC1B,CAAC;CACF;AA9DD,kCA8DC;AAED,MAAM,oBAAoB,GAAG,CAAC,MAAM,EAAE,KAAK,EAAE,KAAK,EAAE,MAAM,CAAU,CAAC","sourcesContent":["import * as fs from \"fs/promises\";\n\nimport { SupportedCodec } from \"../../../../../rtp/src/container/webm\";\nimport {\n  JitterBuffer,\n  MediaStreamTrack,\n  SampleBuilder,\n  WebmOutput,\n} from \"../../..\";\nimport { MediaWriter } from \".\";\n\nexport class WebmFactory extends MediaWriter {\n  webm?: WebmOutput;\n\n  start(tracks: MediaStreamTrack[]) {\n    this.webm = new WebmOutput(\n      fs,\n      \"./test.webm\",\n      tracks.map((track, i) => {\n        const trackNumber = i + 1;\n        const payloadType = track.codec!.payloadType;\n\n        if (track.kind === \"video\") {\n          const codec = ((): SupportedCodec => {\n            switch (track.codec?.name.toLowerCase() as SupportedVideoCodec) {\n              case \"vp8\":\n                return \"VP8\";\n              case \"vp9\":\n                return \"VP9\";\n              case \"h264\":\n                return \"MPEG4/ISO/AVC\";\n              case \"av1x\":\n                return \"AV1\";\n              default:\n                throw new Error();\n            }\n          })();\n          return {\n            kind: \"video\",\n            clockRate: 90000,\n            payloadType,\n            trackNumber,\n            codec,\n            width: this.options.width,\n            height: this.options.height,\n          };\n        } else {\n          return {\n            kind: \"audio\",\n            clockRate: 48000,\n            payloadType,\n            trackNumber,\n            codec: \"OPUS\",\n          };\n        }\n      })\n    );\n\n    tracks.forEach((track) => {\n      const sampleBuilder =\n        track.kind === \"video\"\n          ? new SampleBuilder((h) => !!h.marker).pipe(this.webm!)\n          : new SampleBuilder(() => true).pipe(this.webm!);\n      new JitterBuffer({\n        rtpStream: track.onReceiveRtp,\n        rtcpStream: track.onReceiveRtcp,\n      }).pipe(sampleBuilder);\n    });\n  }\n\n  async stop() {\n    await this.webm!.stop();\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = typeof supportedVideoCodecs[number];\n"]}

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

@@ -1,4 +1,5 @@
 import Event from "rx.mini";
+import { DtlsKeys } from ".";
 import { RTCDataChannel } from "./dataChannel";
 import { EventTarget } from "./helper";
 import { RTCRtpCodecParameters, RTCRtpHeaderExtensionParameters } from "./media/parameters";
@@ -28,7 +29,7 @@ export declare class RTCPeerConnection extends EventTarget {
     readonly iceGatheringStateChange: Event<["new" | "complete" | "gathering"]>;
     readonly iceConnectionStateChange: Event<["disconnected" | "closed" | "completed" | "new" | "connected" | "failed" | "checking"]>;
     readonly signalingStateChange: Event<["closed" | "stable" | "have-local-offer" | "have-remote-offer" | "have-local-pranswer" | "have-remote-pranswer"]>;
-    readonly connectionStateChange: Event<["disconnected" | "closed" | "new" | "connected" | "failed" | "connecting"]>;
+    readonly connectionStateChange: Event<["disconnected" | "closed" | "new" | "connected" | "connecting" | "failed"]>;
     readonly onDataChannel: Event<[RTCDataChannel]>;
     readonly onRemoteTransceiverAdded: Event<[RTCRtpTransceiver]>;
     /**
@@ -47,7 +48,7 @@ export declare class RTCPeerConnection extends EventTarget {
     onconnectionstatechange?: Callback;
     private readonly router;
     private readonly certificates;
-    private sctpRemotePort?;
+    sctpRemotePort?: number;
     private seenMid;
     private currentLocalDescription?;
     private currentRemoteDescription?;
@@ -55,7 +56,7 @@ export declare class RTCPeerConnection extends EventTarget {
     private pendingRemoteDescription?;
     private isClosed;
     private shouldNegotiationneeded;
-    constructor({ codecs, headerExtensions, iceServers, iceTransportPolicy, icePortRange, }?: Partial<PeerConfig>);
+    constructor({ codecs, headerExtensions, iceServers, iceTransportPolicy, icePortRange, dtls, }?: Partial<PeerConfig>);
     get localDescription(): import("./sdp").RTCSessionDescription | undefined;
     get remoteDescription(): import("./sdp").RTCSessionDescription | undefined;
     private get _localDescription();
@@ -94,7 +95,9 @@ export declare class RTCPeerConnection extends EventTarget {
     getTransceivers(): RTCRtpTransceiver[];
     getSenders(): RTCRtpSender[];
     getReceivers(): RTCRtpReceiver[];
-    addTrack(track: MediaStreamTrack, ms?: MediaStream): RTCRtpSender;
+    addTrack(track: MediaStreamTrack, 
+    /**todo impl */
+    ms?: MediaStream): RTCRtpSender;
     createAnswer(): Promise<import("./sdp").RTCSessionDescription>;
     close(): Promise<void>;
     private assertNotClosed;
@@ -121,6 +124,9 @@ export interface PeerConfig {
     iceServers: RTCIceServer[];
     /**Minimum port and Maximum port must not be the same value */
     icePortRange: [number, number] | undefined;
+    dtls: Partial<{
+        keys: DtlsKeys;
+    }>;
 }
 export declare const findCodecByMimeType: (codecs: RTCRtpCodecParameters[], target: RTCRtpCodecParameters) => RTCRtpCodecParameters | undefined;
 export declare type RTCIceServer = {

package/lib/webrtc/src/peerConnection.js

@@ -27,6 +27,7 @@ const debug_1 = __importDefault(require("debug"));
 const lodash_1 = require("lodash");
 const rx_mini_1 = __importDefault(require("rx.mini"));
 const uuid = __importStar(require("uuid"));
+const _1 = require(".");
 const const_1 = require("./const");
 const dataChannel_1 = require("./dataChannel");
 const helper_1 = require("./helper");
@@ -43,7 +44,7 @@ const sctp_1 = require("./transport/sctp");
 const utils_1 = require("./utils");
 const log = (0, debug_1.default)("werift:packages/webrtc/src/peerConnection.ts");
 class RTCPeerConnection extends helper_1.EventTarget {
-    constructor({ codecs, headerExtensions, iceServers, iceTransportPolicy, icePortRange, } = {}) {
+    constructor({ codecs, headerExtensions, iceServers, iceTransportPolicy, icePortRange, dtls, } = {}) {
         super();
         this.cname = uuid.v4();
         this.masterTransportEstablished = false;
@@ -101,15 +102,26 @@ class RTCPeerConnection extends helper_1.EventTarget {
             this.configuration.codecs.audio = codecs.audio;
         if (codecs?.video)
             this.configuration.codecs.video = codecs.video;
-        [
+        for (const [i, codecParams] of (0, helper_1.enumerate)([
             ...(this.configuration.codecs.audio || []),
             ...(this.configuration.codecs.video || []),
-        ].forEach((v, i) => {
-            v.payloadType = 96 + i;
-            if (v.name.toLowerCase() === "rtx") {
-                v.parameters = { apt: v.payloadType - 1 };
+        ])) {
+            codecParams.payloadType = 96 + i;
+            switch (codecParams.name.toLowerCase()) {
+                case "rtx":
+                    {
+                        codecParams.parameters = `apt=${codecParams.payloadType - 1}`;
+                    }
+                    break;
+                case "red":
+                    {
+                        const redundant = codecParams.payloadType + 1;
+                        codecParams.parameters = `${redundant}/${redundant}`;
+                        codecParams.payloadType = 63;
+                    }
+                    break;
             }
-        });
+        }
         if (headerExtensions?.audio)
             this.configuration.headerExtensions.audio = headerExtensions.audio;
         if (headerExtensions?.video)
@@ -120,6 +132,12 @@ class RTCPeerConnection extends helper_1.EventTarget {
         ].forEach((v, i) => {
             v.id = 1 + i;
         });
+        if (dtls) {
+            const { keys } = dtls;
+            if (keys) {
+                this.certificates.push(new dtls_1.RTCCertificate(keys.keyPem, keys.certPem, keys.signatureHash));
+            }
+        }
         this.iceConnectionStateChange.subscribe((state) => {
             switch (state) {
                 case "disconnected":
@@ -441,7 +459,8 @@ class RTCPeerConnection extends helper_1.EventTarget {
             headerExtensions: transceiver.headerExtensions,
             encodings: Object.values(transceiver.codecs.reduce((acc, codec) => {
                 if (codec.name.toLowerCase() === "rtx") {
-                    const apt = acc[codec.parameters["apt"]];
+                    const params = (0, _1.codecParametersFromString)(codec.parameters ?? "");
+                    const apt = acc[params["apt"]];
                     if (apt && media.ssrc.length === 2) {
                         apt.rtx = new parameters_1.RTCRtpRtxParameters({ ssrc: media.ssrc[1].ssrc });
                     }
@@ -486,7 +505,8 @@ class RTCPeerConnection extends helper_1.EventTarget {
                     if (!existCodec)
                         return false;
                     if (existCodec?.name.toLowerCase() === "rtx") {
-                        const pt = existCodec.parameters["apt"];
+                        const params = (0, _1.codecParametersFromString)(existCodec.parameters ?? "");
+                        const pt = params["apt"];
                         const origin = remoteMedia.rtp.codecs.find((c) => c.payloadType === pt);
                         if (!origin)
                             return false;
@@ -666,7 +686,9 @@ class RTCPeerConnection extends helper_1.EventTarget {
         return this.getTransceivers().map((t) => t.receiver);
     }
     // todo fix
-    addTrack(track, ms) {
+    addTrack(track, 
+    /**todo impl */
+    ms) {
         if (this.isClosed)
             throw new Error("is closed");
         if (this.getSenders().find((sender) => sender.track?.uuid === track.uuid)) {
@@ -927,5 +949,6 @@ exports.defaultPeerConfig = {
     iceTransportPolicy: "all",
     iceServers: [{ urls: "stun:stun.l.google.com:19302" }],
     icePortRange: undefined,
+    dtls: {},
 };
 //# sourceMappingURL=peerConnection.js.map