werift 0.21.0 → 0.21.2

package/lib/rtp/src/extra/processor/webmCallback.d.ts

@@ -8,4 +8,9 @@ export declare class WebmCallback extends WebmBase {
     inputVideo: (input: WebmInput) => void;
     destroy: () => void;
 }
-export declare const saveToFileSystem: (path: string) => (value: WebmOutput) => Promise<void>;
+/**
+ *
+ * @param path
+ * @returns eol
+ */
+export declare const saveToFileSystem: (path: string) => (value: WebmOutput) => Promise<boolean | undefined>;

package/lib/rtp/src/extra/processor/webmCallback.js

@@ -61,9 +61,15 @@ class WebmCallback extends webm_1.WebmBase {
     }
 }
 exports.WebmCallback = WebmCallback;
+/**
+ *
+ * @param path
+ * @returns eol
+ */
 const saveToFileSystem = (path) => async (value) => {
     if (value.saveToFile) {
         await (0, promises_1.appendFile)(path, value.saveToFile);
+        return false;
     }
     else if (value.eol) {
         const { durationElement } = value.eol;
@@ -75,6 +81,7 @@ const saveToFileSystem = (path) => async (value) => {
         const resize = (0, webm_1.replaceSegmentSize)(meta.size);
         await handler.write(resize, 0, resize.length, webm_1.SegmentSizePosition);
         await handler.close();
+        return true;
     }
 };
 exports.saveToFileSystem = saveToFileSystem;

package/lib/rtp/src/extra/processor/webmCallback.js.map

@@ -1 +1 @@
-{"version":3,"file":"webmCallback.js","sourceRoot":"","sources":["../../../../../../rtp/src/extra/processor/webmCallback.ts"],"names":[],"mappings":";;;AAAA,0CAAqD;AAErD,6BAAqC;AACrC,iCASgB;AAEhB,MAAa,YAAa,SAAQ,eAAQ;IAGxC,YAAY,MAAmB,EAAE,UAAsB,EAAE;QACvD,KAAK,CACH,MAAM,EACN,KAAK,EAAE,MAAM,EAAE,EAAE;YACf,MAAM,EAAE,GAAG,IAAI,CAAC,EAAE,CAAC;YACnB,IAAI,EAAE,EAAE,CAAC;gBACP,MAAM,IAAI,CAAC,KAAK,CAAC,IAAI,CAAC,GAAG,EAAE,CAAC,EAAE,CAAC,MAAM,CAAC,CAAC,CAAC;YAC1C,CAAC;QACH,CAAC,EACD,OAAO,CACR,CAAC;QAZI;;;;;WAA0C;QAC1C;;;;mBAAQ,IAAI,gBAAY,EAAE;WAAC;QAcnC;;;;mBAAO,CAAC,EAAwC,EAAE,EAAE;gBAClD,IAAI,CAAC,EAAE,GAAG,EAAE,CAAC;gBACb,IAAI,CAAC,KAAK,EAAE,CAAC;YACf,CAAC;WAAC;QAEF;;;;mBAAa,CAAC,KAAgB,EAAE,EAAE;gBAChC,IAAI,CAAC,iBAAiB,CAAC,KAAK,CAAC,CAAC;YAChC,CAAC;WAAC;QACF;;;;mBAAa,CAAC,KAAgB,EAAE,EAAE;gBAChC,IAAI,CAAC,iBAAiB,CAAC,KAAK,CAAC,CAAC;YAChC,CAAC;WAAC;QAEF;;;;mBAAU,GAAG,EAAE;gBACb,IAAI,CAAC,EAAE,GAAG,SAAS,CAAC;gBACpB,IAAI,CAAC,KAAK,CAAC,MAAM,EAAE,CAAC;YACtB,CAAC;WAAC;IAjBF,CAAC;CAkBF;AAhCD,oCAgCC;AAEM,MAAM,gBAAgB,GAAG,CAAC,IAAY,EAAE,EAAE,CAAC,KAAK,EAAE,KAAiB,EAAE,EAAE;IAC5E,IAAI,KAAK,CAAC,UAAU,EAAE,CAAC;QACrB,MAAM,IAAA,qBAAU,EAAC,IAAI,EAAE,KAAK,CAAC,UAAU,CAAC,CAAC;IAC3C,CAAC;SAAM,IAAI,KAAK,CAAC,GAAG,EAAE,CAAC;QACrB,MAAM,EAAE,eAAe,EAAE,GAAG,KAAK,CAAC,GAAG,CAAC;QACtC,MAAM,OAAO,GAAG,MAAM,IAAA,eAAI,EAAC,IAAI,EAAE,IAAI,CAAC,CAAC;QAEvC,eAAe;QACf,MAAM,OAAO,CAAC,KAAK,CACjB,eAAe,EACf,CAAC,EACD,eAAe,CAAC,MAAM,EACtB,uBAAgB,CACjB,CAAC;QAEF,WAAW;QACX,MAAM,IAAI,GAAG,MAAM,IAAA,eAAI,EAAC,IAAI,CAAC,CAAC;QAC9B,MAAM,MAAM,GAAG,IAAA,yBAAkB,EAAC,IAAI,CAAC,IAAI,CAAC,CAAC;QAC7C,MAAM,OAAO,CAAC,KAAK,CAAC,MAAM,EAAE,CAAC,EAAE,MAAM,CAAC,MAAM,EAAE,0BAAmB,CAAC,CAAC;QAEnE,MAAM,OAAO,CAAC,KAAK,EAAE,CAAC;IACxB,CAAC;AACH,CAAC,CAAC;AAtBW,QAAA,gBAAgB,oBAsB3B","sourcesContent":["import { appendFile, open, stat } from \"fs/promises\";\n\nimport { PromiseQueue } from \"../..\";\nimport {\n  DurationPosition,\n  SegmentSizePosition,\n  WebmBase,\n  type WebmInput,\n  type WebmOption,\n  type WebmOutput,\n  type WebmTrack,\n  replaceSegmentSize,\n} from \"./webm\";\n\nexport class WebmCallback extends WebmBase {\n  private cb?: (input: WebmOutput) => Promise<void>;\n  private queue = new PromiseQueue();\n  constructor(tracks: WebmTrack[], options: WebmOption = {}) {\n    super(\n      tracks,\n      async (output) => {\n        const cb = this.cb;\n        if (cb) {\n          await this.queue.push(() => cb(output));\n        }\n      },\n      options,\n    );\n  }\n\n  pipe = (cb: (input: WebmOutput) => Promise<void>) => {\n    this.cb = cb;\n    this.start();\n  };\n\n  inputAudio = (input: WebmInput) => {\n    this.processAudioInput(input);\n  };\n  inputVideo = (input: WebmInput) => {\n    this.processVideoInput(input);\n  };\n\n  destroy = () => {\n    this.cb = undefined;\n    this.queue.cancel();\n  };\n}\n\nexport const saveToFileSystem = (path: string) => async (value: WebmOutput) => {\n  if (value.saveToFile) {\n    await appendFile(path, value.saveToFile);\n  } else if (value.eol) {\n    const { durationElement } = value.eol;\n    const handler = await open(path, \"r+\");\n\n    // set duration\n    await handler.write(\n      durationElement,\n      0,\n      durationElement.length,\n      DurationPosition,\n    );\n\n    // set size\n    const meta = await stat(path);\n    const resize = replaceSegmentSize(meta.size);\n    await handler.write(resize, 0, resize.length, SegmentSizePosition);\n\n    await handler.close();\n  }\n};\n"]}
+{"version":3,"file":"webmCallback.js","sourceRoot":"","sources":["../../../../../../rtp/src/extra/processor/webmCallback.ts"],"names":[],"mappings":";;;AAAA,0CAAqD;AAErD,6BAAqC;AACrC,iCASgB;AAEhB,MAAa,YAAa,SAAQ,eAAQ;IAGxC,YAAY,MAAmB,EAAE,UAAsB,EAAE;QACvD,KAAK,CACH,MAAM,EACN,KAAK,EAAE,MAAM,EAAE,EAAE;YACf,MAAM,EAAE,GAAG,IAAI,CAAC,EAAE,CAAC;YACnB,IAAI,EAAE,EAAE,CAAC;gBACP,MAAM,IAAI,CAAC,KAAK,CAAC,IAAI,CAAC,GAAG,EAAE,CAAC,EAAE,CAAC,MAAM,CAAC,CAAC,CAAC;YAC1C,CAAC;QACH,CAAC,EACD,OAAO,CACR,CAAC;QAZI;;;;;WAA0C;QAC1C;;;;mBAAQ,IAAI,gBAAY,EAAE;WAAC;QAcnC;;;;mBAAO,CAAC,EAAwC,EAAE,EAAE;gBAClD,IAAI,CAAC,EAAE,GAAG,EAAE,CAAC;gBACb,IAAI,CAAC,KAAK,EAAE,CAAC;YACf,CAAC;WAAC;QAEF;;;;mBAAa,CAAC,KAAgB,EAAE,EAAE;gBAChC,IAAI,CAAC,iBAAiB,CAAC,KAAK,CAAC,CAAC;YAChC,CAAC;WAAC;QACF;;;;mBAAa,CAAC,KAAgB,EAAE,EAAE;gBAChC,IAAI,CAAC,iBAAiB,CAAC,KAAK,CAAC,CAAC;YAChC,CAAC;WAAC;QAEF;;;;mBAAU,GAAG,EAAE;gBACb,IAAI,CAAC,EAAE,GAAG,SAAS,CAAC;gBACpB,IAAI,CAAC,KAAK,CAAC,MAAM,EAAE,CAAC;YACtB,CAAC;WAAC;IAjBF,CAAC;CAkBF;AAhCD,oCAgCC;AAED;;;;GAIG;AACI,MAAM,gBAAgB,GAAG,CAAC,IAAY,EAAE,EAAE,CAAC,KAAK,EAAE,KAAiB,EAAE,EAAE;IAC5E,IAAI,KAAK,CAAC,UAAU,EAAE,CAAC;QACrB,MAAM,IAAA,qBAAU,EAAC,IAAI,EAAE,KAAK,CAAC,UAAU,CAAC,CAAC;QAEzC,OAAO,KAAK,CAAC;IACf,CAAC;SAAM,IAAI,KAAK,CAAC,GAAG,EAAE,CAAC;QACrB,MAAM,EAAE,eAAe,EAAE,GAAG,KAAK,CAAC,GAAG,CAAC;QACtC,MAAM,OAAO,GAAG,MAAM,IAAA,eAAI,EAAC,IAAI,EAAE,IAAI,CAAC,CAAC;QAEvC,eAAe;QACf,MAAM,OAAO,CAAC,KAAK,CACjB,eAAe,EACf,CAAC,EACD,eAAe,CAAC,MAAM,EACtB,uBAAgB,CACjB,CAAC;QAEF,WAAW;QACX,MAAM,IAAI,GAAG,MAAM,IAAA,eAAI,EAAC,IAAI,CAAC,CAAC;QAC9B,MAAM,MAAM,GAAG,IAAA,yBAAkB,EAAC,IAAI,CAAC,IAAI,CAAC,CAAC;QAC7C,MAAM,OAAO,CAAC,KAAK,CAAC,MAAM,EAAE,CAAC,EAAE,MAAM,CAAC,MAAM,EAAE,0BAAmB,CAAC,CAAC;QAEnE,MAAM,OAAO,CAAC,KAAK,EAAE,CAAC;QAEtB,OAAO,IAAI,CAAC;IACd,CAAC;AACH,CAAC,CAAC;AA1BW,QAAA,gBAAgB,oBA0B3B","sourcesContent":["import { appendFile, open, stat } from \"fs/promises\";\n\nimport { PromiseQueue } from \"../..\";\nimport {\n  DurationPosition,\n  SegmentSizePosition,\n  WebmBase,\n  type WebmInput,\n  type WebmOption,\n  type WebmOutput,\n  type WebmTrack,\n  replaceSegmentSize,\n} from \"./webm\";\n\nexport class WebmCallback extends WebmBase {\n  private cb?: (input: WebmOutput) => Promise<void>;\n  private queue = new PromiseQueue();\n  constructor(tracks: WebmTrack[], options: WebmOption = {}) {\n    super(\n      tracks,\n      async (output) => {\n        const cb = this.cb;\n        if (cb) {\n          await this.queue.push(() => cb(output));\n        }\n      },\n      options,\n    );\n  }\n\n  pipe = (cb: (input: WebmOutput) => Promise<void>) => {\n    this.cb = cb;\n    this.start();\n  };\n\n  inputAudio = (input: WebmInput) => {\n    this.processAudioInput(input);\n  };\n  inputVideo = (input: WebmInput) => {\n    this.processVideoInput(input);\n  };\n\n  destroy = () => {\n    this.cb = undefined;\n    this.queue.cancel();\n  };\n}\n\n/**\n *\n * @param path\n * @returns eol\n */\nexport const saveToFileSystem = (path: string) => async (value: WebmOutput) => {\n  if (value.saveToFile) {\n    await appendFile(path, value.saveToFile);\n\n    return false;\n  } else if (value.eol) {\n    const { durationElement } = value.eol;\n    const handler = await open(path, \"r+\");\n\n    // set duration\n    await handler.write(\n      durationElement,\n      0,\n      durationElement.length,\n      DurationPosition,\n    );\n\n    // set size\n    const meta = await stat(path);\n    const resize = replaceSegmentSize(meta.size);\n    await handler.write(resize, 0, resize.length, SegmentSizePosition);\n\n    await handler.close();\n\n    return true;\n  }\n};\n"]}

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

@@ -17,3 +17,4 @@ export * from "./transport/ice";
 export * from "./transport/sctp";
 export * from "./types/domain";
 export * from "./utils";
+export * from "./media/codec";

package/lib/webrtc/src/index.js

@@ -33,4 +33,5 @@ __exportStar(require("./transport/ice"), exports);
 __exportStar(require("./transport/sctp"), exports);
 __exportStar(require("./types/domain"), exports);
 __exportStar(require("./utils"), exports);
+__exportStar(require("./media/codec"), exports);
 //# sourceMappingURL=index.js.map

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

@@ -1 +1 @@
-{"version":3,"file":"index.js","sourceRoot":"","sources":["../../../src/index.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;AAAA,mDAAiC;AACjC,8DAA4C;AAC5C,gDAA8B;AAC9B,gDAA8B;AAC9B,gDAA8B;AAC9B,iEAA+C;AAC/C,iEAA+C;AAC/C,qDAAmC;AACnC,sDAAoC;AACpC,oDAAkC;AAClC,yDAAuC;AACvC,gDAA8B;AAC9B,mDAAiC;AACjC,wCAAsB;AACtB,mDAAiC;AACjC,kDAAgC;AAChC,mDAAiC;AACjC,iDAA+B;AAC/B,0CAAwB","sourcesContent":["export * from \"../../common/src\";\nexport * from \"../../dtls/src/cipher/const\";\nexport * from \"../../ice/src\";\nexport * from \"../../rtp/src\";\nexport * from \"./dataChannel\";\nexport * from \"./media/extension/rtcpFeedback\";\nexport * from \"./media/extension/rtpExtension\";\nexport * from \"./media/parameters\";\nexport * from \"./media/rtpReceiver\";\nexport * from \"./media/rtpSender\";\nexport * from \"./media/rtpTransceiver\";\nexport * from \"./media/track\";\nexport * from \"./peerConnection\";\nexport * from \"./sdp\";\nexport * from \"./transport/dtls\";\nexport * from \"./transport/ice\";\nexport * from \"./transport/sctp\";\nexport * from \"./types/domain\";\nexport * from \"./utils\";\n"]}
+{"version":3,"file":"index.js","sourceRoot":"","sources":["../../../src/index.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;AAAA,mDAAiC;AACjC,8DAA4C;AAC5C,gDAA8B;AAC9B,gDAA8B;AAC9B,gDAA8B;AAC9B,iEAA+C;AAC/C,iEAA+C;AAC/C,qDAAmC;AACnC,sDAAoC;AACpC,oDAAkC;AAClC,yDAAuC;AACvC,gDAA8B;AAC9B,mDAAiC;AACjC,wCAAsB;AACtB,mDAAiC;AACjC,kDAAgC;AAChC,mDAAiC;AACjC,iDAA+B;AAC/B,0CAAwB;AACxB,gDAA8B","sourcesContent":["export * from \"../../common/src\";\nexport * from \"../../dtls/src/cipher/const\";\nexport * from \"../../ice/src\";\nexport * from \"../../rtp/src\";\nexport * from \"./dataChannel\";\nexport * from \"./media/extension/rtcpFeedback\";\nexport * from \"./media/extension/rtpExtension\";\nexport * from \"./media/parameters\";\nexport * from \"./media/rtpReceiver\";\nexport * from \"./media/rtpSender\";\nexport * from \"./media/rtpTransceiver\";\nexport * from \"./media/track\";\nexport * from \"./peerConnection\";\nexport * from \"./sdp\";\nexport * from \"./transport/dtls\";\nexport * from \"./transport/ice\";\nexport * from \"./transport/sctp\";\nexport * from \"./types/domain\";\nexport * from \"./utils\";\nexport * from \"./media/codec\";\n"]}

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

@@ -0,0 +1,3 @@
+import { RTCRtpCodecParameters } from "..";
+export declare const useH264: (props?: Partial<RTCRtpCodecParameters>) => RTCRtpCodecParameters;
+export declare const useVP8: (props?: Partial<RTCRtpCodecParameters>) => RTCRtpCodecParameters;

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

@@ -0,0 +1,28 @@
+"use strict";
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.useVP8 = exports.useH264 = void 0;
+const __1 = require("..");
+const useH264 = (props = {}) => new __1.RTCRtpCodecParameters({
+    mimeType: "video/h264",
+    clockRate: 90000,
+    rtcpFeedback: [
+        { type: "nack" },
+        { type: "nack", parameter: "pli" },
+        { type: "goog-remb" },
+    ],
+    parameters: "profile-level-id=42e01f;packetization-mode=1;level-asymmetry-allowed=1",
+    ...props,
+});
+exports.useH264 = useH264;
+const useVP8 = (props = {}) => new __1.RTCRtpCodecParameters({
+    mimeType: "video/VP8",
+    clockRate: 90000,
+    rtcpFeedback: [
+        { type: "nack" },
+        { type: "nack", parameter: "pli" },
+        { type: "goog-remb" },
+    ],
+    ...props,
+});
+exports.useVP8 = useVP8;
+//# sourceMappingURL=codec.js.map

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

@@ -0,0 +1 @@
+{"version":3,"file":"codec.js","sourceRoot":"","sources":["../../../../src/media/codec.ts"],"names":[],"mappings":";;;AAAA,0BAA2C;AAEpC,MAAM,OAAO,GAAG,CAAC,QAAwC,EAAE,EAAE,EAAE,CACpE,IAAI,yBAAqB,CAAC;IACxB,QAAQ,EAAE,YAAY;IACtB,SAAS,EAAE,KAAK;IAChB,YAAY,EAAE;QACZ,EAAE,IAAI,EAAE,MAAM,EAAE;QAChB,EAAE,IAAI,EAAE,MAAM,EAAE,SAAS,EAAE,KAAK,EAAE;QAClC,EAAE,IAAI,EAAE,WAAW,EAAE;KACtB;IACD,UAAU,EACR,wEAAwE;IAC1E,GAAG,KAAK;CACT,CAAC,CAAC;AAZQ,QAAA,OAAO,WAYf;AAEE,MAAM,MAAM,GAAG,CAAC,QAAwC,EAAE,EAAE,EAAE,CACnE,IAAI,yBAAqB,CAAC;IACxB,QAAQ,EAAE,WAAW;IACrB,SAAS,EAAE,KAAK;IAChB,YAAY,EAAE;QACZ,EAAE,IAAI,EAAE,MAAM,EAAE;QAChB,EAAE,IAAI,EAAE,MAAM,EAAE,SAAS,EAAE,KAAK,EAAE;QAClC,EAAE,IAAI,EAAE,WAAW,EAAE;KACtB;IACD,GAAG,KAAK;CACT,CAAC,CAAC;AAVQ,QAAA,MAAM,UAUd","sourcesContent":["import { RTCRtpCodecParameters } from \"..\";\n\nexport const useH264 = (props: Partial<RTCRtpCodecParameters> = {}) =>\n  new RTCRtpCodecParameters({\n    mimeType: \"video/h264\",\n    clockRate: 90000,\n    rtcpFeedback: [\n      { type: \"nack\" },\n      { type: \"nack\", parameter: \"pli\" },\n      { type: \"goog-remb\" },\n    ],\n    parameters:\n      \"profile-level-id=42e01f;packetization-mode=1;level-asymmetry-allowed=1\",\n    ...props,\n  });\n\nexport const useVP8 = (props: Partial<RTCRtpCodecParameters> = {}) =>\n  new RTCRtpCodecParameters({\n    mimeType: \"video/VP8\",\n    clockRate: 90000,\n    rtcpFeedback: [\n      { type: \"nack\" },\n      { type: \"nack\", parameter: \"pli\" },\n      { type: \"goog-remb\" },\n    ],\n    ...props,\n  });\n"]}

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

@@ -5,7 +5,7 @@ import type { RTCDtlsTransport } from "../transport/dtls";
 import type { Kind } from "../types/domain";
 import type { RTCRtpReceiveParameters } from "./parameters";
 import { ReceiverTWCC } from "./receiver/receiverTwcc";
-import type { MediaStreamTrack } from "./track";
+import { MediaStreamTrack } from "./track";
 export declare class RTCRtpReceiver {
     readonly config: PeerConfig;
     kind: Kind;
@@ -34,7 +34,7 @@ export declare class RTCRtpReceiver {
     readonly receiveLastSRTimestamp: {
         [ssrc: number]: number;
     };
-    readonly onPacketLost: Event<[import("werift-rtp/src/rtcp/rtpfb/nack").GenericNack]>;
+    readonly onPacketLost: Event<[import("..").GenericNack]>;
     readonly onRtcp: Event<[RtcpPacket]>;
     dtlsTransport: RTCDtlsTransport;
     sdesMid?: string;

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

@@ -12,6 +12,7 @@ const utils_1 = require("../utils");
 const nack_1 = require("./receiver/nack");
 const receiverTwcc_1 = require("./receiver/receiverTwcc");
 const statistics_1 = require("./receiver/statistics");
+const track_1 = require("./track");
 const log = (0, debug_1.debug)("werift:packages/webrtc/src/media/rtpReceiver.ts");
 class RTCRtpReceiver {
     get codecArray() {
@@ -211,7 +212,7 @@ class RTCRtpReceiver {
     }
     // todo fix
     get track() {
-        return this.tracks[0];
+        return this.tracks[0] ?? new track_1.MediaStreamTrack({ kind: this.kind });
     }
     get nackEnabled() {
         return this.codecArray[0]?.rtcpFeedback.find((f) => f.type === "nack");

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

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

package/lib/webrtc/src/nonstandard/navigator.d.ts

@@ -17,9 +17,10 @@ export declare class MediaDevices extends EventTarget {
     });
     readonly getUserMedia: (constraints: MediaStreamConstraints) => Promise<MediaStream>;
     readonly getDisplayMedia: (constraints: MediaStreamConstraints) => Promise<MediaStream>;
-    readonly getUdpMedia: ({ port, codec, }: {
+    readonly getUdpMedia: ({ port, codec, kind, }: {
         port: number;
         codec: ConstructorParameters<typeof RTCRtpCodecParameters>[0];
+        kind: "audio" | "video";
     }) => {
         track: MediaStreamTrack;
         disposer: () => void;
@@ -78,4 +79,5 @@ interface DoubleRange {
     max?: number;
     min?: number;
 }
+export declare const navigator: Navigator;
 export {};

package/lib/webrtc/src/nonstandard/navigator.js

@@ -1,6 +1,6 @@
 "use strict";
 Object.defineProperty(exports, "__esModule", { value: true });
-exports.MediaDevices = exports.Navigator = void 0;
+exports.navigator = exports.MediaDevices = exports.Navigator = void 0;
 const crypto_1 = require("crypto");
 const dgram_1 = require("dgram");
 const jspack_1 = require("@shinyoshiaki/jspack");
@@ -86,9 +86,9 @@ class MediaDevices extends EventTarget {
             enumerable: true,
             configurable: true,
             writable: true,
-            value: ({ port, codec, }) => {
+            value: ({ port, codec, kind, }) => {
                 const track = new track_1.MediaStreamTrack({
-                    kind: "audio",
+                    kind,
                     codec: new __1.RTCRtpCodecParameters(codec),
                 });
                 const udp = (0, dgram_1.createSocket)("udp4");
@@ -107,4 +107,5 @@ class MediaDevices extends EventTarget {
     }
 }
 exports.MediaDevices = MediaDevices;
+exports.navigator = new Navigator();
 //# sourceMappingURL=navigator.js.map

package/lib/webrtc/src/nonstandard/navigator.js.map

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new MediaStreamTrack({ kind: \"video\" })\n      : undefined;\n    if (video) {\n      this.video?.onReceiveRtp.subscribe((rtp) => {\n        const cloned = rtp.clone();\n        cloned.header.ssrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n        video.onReceiveRtp.execute(cloned);\n      });\n    }\n    const audio = constraints.audio\n      ? new MediaStreamTrack({ kind: \"audio\" })\n      : undefined;\n    if (audio) {\n      this.audio?.onReceiveRtp.subscribe((rtp) => {\n        const cloned = rtp.clone();\n        cloned.header.ssrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n        audio.onReceiveRtp.execute(cloned);\n      });\n    }\n\n    if (constraints.video && constraints.audio) {\n      return new MediaStream([video!, audio!]);\n    } else if (constraints.audio) {\n      return new MediaStream([audio!]);\n    } else if (constraints.video) {\n      return new MediaStream([video!]);\n    }\n\n    throw new Error(\"Not implemented\");\n  };\n\n  readonly getDisplayMedia = this.getUserMedia;\n\n  readonly getUdpMedia = ({\n    port,\n    codec,\n  }: {\n    port: number;\n    codec: ConstructorParameters<typeof RTCRtpCodecParameters>[0];\n  }) => {\n    const track = new MediaStreamTrack({\n      kind: \"audio\",\n      codec: new RTCRtpCodecParameters(codec),\n    });\n\n    const udp = createSocket(\"udp4\");\n    udp.bind(port);\n    udp.on(\"message\", (data) => {\n      track.writeRtp(data);\n    });\n\n    const disposer = () => {\n      udp.close();\n    };\n\n    return { track, disposer };\n  };\n}\n\ninterface MediaStreamConstraints {\n  audio?: boolean | MediaTrackConstraints;\n  peerIdentity?: string;\n  preferCurrentTab?: boolean;\n  video?: boolean | MediaTrackConstraints;\n}\n\ninterface MediaTrackConstraints extends MediaTrackConstraintSet {\n  advanced?: MediaTrackConstraintSet[];\n}\n\ninterface MediaTrackConstraintSet {\n  aspectRatio?: ConstrainDouble;\n  autoGainControl?: ConstrainBoolean;\n  channelCount?: ConstrainULong;\n  deviceId?: ConstrainDOMString;\n  displaySurface?: ConstrainDOMString;\n  echoCancellation?: ConstrainBoolean;\n  facingMode?: ConstrainDOMString;\n  frameRate?: ConstrainDouble;\n  groupId?: ConstrainDOMString;\n  height?: ConstrainULong;\n  noiseSuppression?: ConstrainBoolean;\n  sampleRate?: ConstrainULong;\n  sampleSize?: ConstrainULong;\n  width?: ConstrainULong;\n}\n\ntype ConstrainDOMString = string | string[] | ConstrainDOMStringParameters;\ninterface ConstrainDOMStringParameters {\n  exact?: string | string[];\n  ideal?: string | string[];\n}\ntype ConstrainBoolean = boolean | ConstrainBooleanParameters;\ninterface ConstrainBooleanParameters {\n  exact?: boolean;\n  ideal?: boolean;\n}\ntype ConstrainULong = number | ConstrainULongRange;\ninterface ConstrainULongRange extends ULongRange {\n  exact?: number;\n  ideal?: number;\n}\ninterface ULongRange {\n  max?: number;\n  min?: number;\n}\ntype ConstrainDouble = number | ConstrainDoubleRange;\ninterface ConstrainDoubleRange extends DoubleRange {\n  exact?: number;\n  ideal?: number;\n}\ninterface DoubleRange {\n  max?: number;\n  min?: number;\n}\n"]}
+{"version":3,"file":"navigator.js","sourceRoot":"","sources":["../../../../src/nonstandard/navigator.ts"],"names":[],"mappings":";;;AAAA,mCAAqC;AACrC,iCAAqC;AACrC,iDAA8C;AAC9C,0BAA2C;AAC3C,0CAA+D;AAE/D,MAAa,SAAS;IAGpB,YAAY,QAAuD,EAAE;QAFrE;;;;;WAA2B;QAGzB,IAAI,CAAC,YAAY,GAAG,IAAI,YAAY,CAAC,KAAK,CAAC,CAAC;IAC9C,CAAC;CACF;AAND,8BAMC;AAED,MAAa,YAAa,SAAQ,WAAW;IAI3C,YACW,KAA6D;QAEtE,KAAK,EAAE,CAAC;QAFR;;;;mBAAS,KAAK;WAAwD;QAJxE;;;;;WAAyB;QACzB;;;;;WAAyB;QAUhB;;;;mBAAe,KAAK,EAC3B,WAAmC,EACb,EAAE;gBACxB,MAAM,KAAK,GAAG,WAAW,CAAC,KAAK;oBAC7B,CAAC,CAAC,IAAI,wBAAgB,CAAC,EAAE,IAAI,EAAE,OAAO,EAAE,CAAC;oBACzC,CAAC,CAAC,SAAS,CAAC;gBACd,IAAI,KAAK,EAAE,CAAC;oBACV,IAAI,CAAC,KAAK,EAAE,YAAY,CAAC,SAAS,CAAC,CAAC,GAAG,EAAE,EAAE;wBACzC,MAAM,MAAM,GAAG,GAAG,CAAC,KAAK,EAAE,CAAC;wBAC3B,MAAM,CAAC,MAAM,CAAC,IAAI,GAAG,eAAM,CAAC,MAAM,CAAC,IAAI,EAAE,IAAA,oBAAW,EAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;wBAC5D,KAAK,CAAC,YAAY,CAAC,OAAO,CAAC,MAAM,CAAC,CAAC;oBACrC,CAAC,CAAC,CAAC;gBACL,CAAC;gBACD,MAAM,KAAK,GAAG,WAAW,CAAC,KAAK;oBAC7B,CAAC,CAAC,IAAI,wBAAgB,CAAC,EAAE,IAAI,EAAE,OAAO,EAAE,CAAC;oBACzC,CAAC,CAAC,SAAS,CAAC;gBACd,IAAI,KAAK,EAAE,CAAC;oBACV,IAAI,CAAC,KAAK,EAAE,YAAY,CAAC,SAAS,CAAC,CAAC,GAAG,EAAE,EAAE;wBACzC,MAAM,MAAM,GAAG,GAAG,CAAC,KAAK,EAAE,CAAC;wBAC3B,MAAM,CAAC,MAAM,CAAC,IAAI,GAAG,eAAM,CAAC,MAAM,CAAC,IAAI,EAAE,IAAA,oBAAW,EAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;wBAC5D,KAAK,CAAC,YAAY,CAAC,OAAO,CAAC,MAAM,CAAC,CAAC;oBACrC,CAAC,CAAC,CAAC;gBACL,CAAC;gBAED,IAAI,WAAW,CAAC,KAAK,IAAI,WAAW,CAAC,KAAK,EAAE,CAAC;oBAC3C,OAAO,IAAI,mBAAW,CAAC,CAAC,KAAM,EAAE,KAAM,CAAC,CAAC,CAAC;gBAC3C,CAAC;qBAAM,IAAI,WAAW,CAAC,KAAK,EAAE,CAAC;oBAC7B,OAAO,IAAI,mBAAW,CAAC,CAAC,KAAM,CAAC,CAAC,CAAC;gBACnC,CAAC;qBAAM,IAAI,WAAW,CAAC,KAAK,EAAE,CAAC;oBAC7B,OAAO,IAAI,mBAAW,CAAC,CAAC,KAAM,CAAC,CAAC,CAAC;gBACnC,CAAC;gBAED,MAAM,IAAI,KAAK,CAAC,iBAAiB,CAAC,CAAC;YACrC,CAAC;WAAC;QAEO;;;;mBAAkB,IAAI,CAAC,YAAY;WAAC;QAEpC;;;;mBAAc,CAAC,EACtB,IAAI,EACJ,KAAK,EACL,IAAI,GAKL,EAAE,EAAE;gBACH,MAAM,KAAK,GAAG,IAAI,wBAAgB,CAAC;oBACjC,IAAI;oBACJ,KAAK,EAAE,IAAI,yBAAqB,CAAC,KAAK,CAAC;iBACxC,CAAC,CAAC;gBAEH,MAAM,GAAG,GAAG,IAAA,oBAAY,EAAC,MAAM,CAAC,CAAC;gBACjC,GAAG,CAAC,IAAI,CAAC,IAAI,CAAC,CAAC;gBACf,GAAG,CAAC,EAAE,CAAC,SAAS,EAAE,CAAC,IAAI,EAAE,EAAE;oBACzB,KAAK,CAAC,QAAQ,CAAC,IAAI,CAAC,CAAC;gBACvB,CAAC,CAAC,CAAC;gBAEH,MAAM,QAAQ,GAAG,GAAG,EAAE;oBACpB,GAAG,CAAC,KAAK,EAAE,CAAC;gBACd,CAAC,CAAC;gBAEF,OAAO,EAAE,KAAK,EAAE,QAAQ,EAAE,CAAC;YAC7B,CAAC;WAAC;QAlEA,IAAI,CAAC,KAAK,GAAG,KAAK,CAAC,KAAK,CAAC;QACzB,IAAI,CAAC,KAAK,GAAG,KAAK,CAAC,KAAK,CAAC;IAC3B,CAAC;CAiEF;AA3ED,oCA2EC;AA2DY,QAAA,SAAS,GAAG,IAAI,SAAS,EAAE,CAAC","sourcesContent":["import { randomBytes } from \"crypto\";\nimport { createSocket } from \"dgram\";\nimport { jspack } from \"@shinyoshiaki/jspack\";\nimport { RTCRtpCodecParameters } from \"..\";\nimport { MediaStream, MediaStreamTrack } from \"../media/track\";\n\nexport class Navigator {\n  mediaDevices: MediaDevices;\n\n  constructor(props: ConstructorParameters<typeof MediaDevices>[0] = {}) {\n    this.mediaDevices = new MediaDevices(props);\n  }\n}\n\nexport class MediaDevices extends EventTarget {\n  video?: MediaStreamTrack;\n  audio?: MediaStreamTrack;\n\n  constructor(\n    readonly props: { video?: MediaStreamTrack; audio?: MediaStreamTrack },\n  ) {\n    super();\n    this.video = props.video;\n    this.audio = props.audio;\n  }\n\n  readonly getUserMedia = async (\n    constraints: MediaStreamConstraints,\n  ): Promise<MediaStream> => {\n    const video = constraints.video\n      ? new MediaStreamTrack({ kind: \"video\" })\n      : undefined;\n    if (video) {\n      this.video?.onReceiveRtp.subscribe((rtp) => {\n        const cloned = rtp.clone();\n        cloned.header.ssrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n        video.onReceiveRtp.execute(cloned);\n      });\n    }\n    const audio = constraints.audio\n      ? new MediaStreamTrack({ kind: \"audio\" })\n      : undefined;\n    if (audio) {\n      this.audio?.onReceiveRtp.subscribe((rtp) => {\n        const cloned = rtp.clone();\n        cloned.header.ssrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n        audio.onReceiveRtp.execute(cloned);\n      });\n    }\n\n    if (constraints.video && constraints.audio) {\n      return new MediaStream([video!, audio!]);\n    } else if (constraints.audio) {\n      return new MediaStream([audio!]);\n    } else if (constraints.video) {\n      return new MediaStream([video!]);\n    }\n\n    throw new Error(\"Not implemented\");\n  };\n\n  readonly getDisplayMedia = this.getUserMedia;\n\n  readonly getUdpMedia = ({\n    port,\n    codec,\n    kind,\n  }: {\n    port: number;\n    codec: ConstructorParameters<typeof RTCRtpCodecParameters>[0];\n    kind: \"audio\" | \"video\";\n  }) => {\n    const track = new MediaStreamTrack({\n      kind,\n      codec: new RTCRtpCodecParameters(codec),\n    });\n\n    const udp = createSocket(\"udp4\");\n    udp.bind(port);\n    udp.on(\"message\", (data) => {\n      track.writeRtp(data);\n    });\n\n    const disposer = () => {\n      udp.close();\n    };\n\n    return { track, disposer };\n  };\n}\n\ninterface MediaStreamConstraints {\n  audio?: boolean | MediaTrackConstraints;\n  peerIdentity?: string;\n  preferCurrentTab?: boolean;\n  video?: boolean | MediaTrackConstraints;\n}\n\ninterface MediaTrackConstraints extends MediaTrackConstraintSet {\n  advanced?: MediaTrackConstraintSet[];\n}\n\ninterface MediaTrackConstraintSet {\n  aspectRatio?: ConstrainDouble;\n  autoGainControl?: ConstrainBoolean;\n  channelCount?: ConstrainULong;\n  deviceId?: ConstrainDOMString;\n  displaySurface?: ConstrainDOMString;\n  echoCancellation?: ConstrainBoolean;\n  facingMode?: ConstrainDOMString;\n  frameRate?: ConstrainDouble;\n  groupId?: ConstrainDOMString;\n  height?: ConstrainULong;\n  noiseSuppression?: ConstrainBoolean;\n  sampleRate?: ConstrainULong;\n  sampleSize?: ConstrainULong;\n  width?: ConstrainULong;\n}\n\ntype ConstrainDOMString = string | string[] | ConstrainDOMStringParameters;\ninterface ConstrainDOMStringParameters {\n  exact?: string | string[];\n  ideal?: string | string[];\n}\ntype ConstrainBoolean = boolean | ConstrainBooleanParameters;\ninterface ConstrainBooleanParameters {\n  exact?: boolean;\n  ideal?: boolean;\n}\ntype ConstrainULong = number | ConstrainULongRange;\ninterface ConstrainULongRange extends ULongRange {\n  exact?: number;\n  ideal?: number;\n}\ninterface ULongRange {\n  max?: number;\n  min?: number;\n}\ntype ConstrainDouble = number | ConstrainDoubleRange;\ninterface ConstrainDoubleRange extends DoubleRange {\n  exact?: number;\n  ideal?: number;\n}\ninterface DoubleRange {\n  max?: number;\n  min?: number;\n}\n\nexport const navigator = new Navigator();\n"]}

package/lib/webrtc/src/nonstandard/recorder/writer/webm.d.ts

@@ -4,6 +4,8 @@ import { type MediaStreamTrack } from "../../..";
 import { RtpSourceCallback } from "../../../../../rtp/src/extra";
 export declare class WebmFactory extends MediaWriter {
     rtpSources: RtpSourceCallback[];
+    private onEol;
+    private ended;
     unSubscribers: EventDisposer;
     start(tracks: MediaStreamTrack[]): Promise<void>;
     stop(): Promise<void>;

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

@@ -16,6 +16,18 @@ class WebmFactory extends _1.MediaWriter {
             writable: true,
             value: []
         });
+        Object.defineProperty(this, "onEol", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: new common_1.Event()
+        });
+        Object.defineProperty(this, "ended", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: false
+        });
         Object.defineProperty(this, "unSubscribers", {
             enumerable: true,
             configurable: true,
@@ -130,7 +142,13 @@ class WebmFactory extends _1.MediaWriter {
             return rtpSource;
         });
         if (this.props.path) {
-            webm.pipe((0, extra_1.saveToFileSystem)(this.props.path));
+            webm.pipe(async (o) => {
+                const eol = await (0, extra_1.saveToFileSystem)(this.props.path)(o);
+                if (eol) {
+                    this.onEol.execute();
+                    this.ended = true;
+                }
+            });
         }
         else if (this.props.stream) {
             webm.pipe(async (o) => {
@@ -140,6 +158,9 @@ class WebmFactory extends _1.MediaWriter {
     }
     async stop() {
         await Promise.all(this.rtpSources.map((r) => r.stop()));
+        if (!this.ended) {
+            await this.onEol.asPromise(5000);
+        }
         this.unSubscribers.dispose();
     }
 }

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

@@ -1 +1 @@
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{ unlink } from \"fs/promises\";\nimport { EventDisposer } from \"../../../imports/common\";\n\nimport { MediaWriter } from \".\";\nimport { type MediaStreamTrack, WeriftError } from \"../../..\";\nimport {\n  DepacketizeCallback,\n  JitterBufferCallback,\n  LipsyncCallback,\n  NtpTimeCallback,\n  RtcpSourceCallback,\n  RtpSourceCallback,\n  RtpTimeCallback,\n  type SupportedCodec,\n  WebmCallback,\n  type WebmTrack,\n  saveToFileSystem,\n} from \"../../../../../rtp/src/extra\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n  rtpSources: RtpSourceCallback[] = [];\n\n  unSubscribers = new EventDisposer();\n\n  async start(tracks: MediaStreamTrack[]) {\n    if (this.props.path) {\n      await unlink(this.props.path).catch((e) => e);\n    }\n\n    const inputTracks: (WebmTrack & { track: MediaStreamTrack })[] = tracks.map(\n      (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 WeriftError({\n                  message: \"unsupported codec\",\n                  payload: { track, path: sourcePath },\n                });\n            }\n          })();\n          return {\n            kind: \"video\" as const,\n            codec,\n            clockRate: 90000,\n            trackNumber,\n            width: this.props.width ?? 640,\n            height: this.props.height ?? 360,\n            roll: this.props.roll,\n            payloadType,\n            track,\n          };\n        } else {\n          return {\n            kind: \"audio\" as const,\n            codec: \"OPUS\" as const,\n            clockRate: 48000,\n            trackNumber,\n            payloadType,\n            track,\n          };\n        }\n      },\n    );\n\n    const webm = new WebmCallback(inputTracks, {\n      duration: this.props.defaultDuration ?? 1000 * 60 * 60 * 24,\n    });\n    const lipsync = new LipsyncCallback();\n\n    if (inputTracks.length === 1 || this.props.disableNtp) {\n      this.props.disableLipSync = true;\n    }\n\n    this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {\n      const rtpSource = new RtpSourceCallback();\n      const rtcpSource = new RtcpSourceCallback();\n      track.onReceiveRtp\n        .subscribe((rtp) => {\n          rtpSource.input(rtp.clone());\n        })\n        .disposer(this.unSubscribers);\n      track.onReceiveRtcp\n        .subscribe((rtcp) => {\n          rtcpSource.input(rtcp);\n        })\n        .disposer(this.unSubscribers);\n      const time = this.props.disableNtp\n        ? new RtpTimeCallback(clockRate)\n        : new NtpTimeCallback(clockRate);\n\n      if (track.kind === \"video\") {\n        const depacketizer = new DepacketizeCallback(codec, {\n          isFinalPacketInSequence: (h) => h.marker,\n        });\n        const jitterBuffer = new JitterBufferCallback(clockRate, {\n          bufferSize: this.props.jitterBufferSize,\n          latency: this.props.jitterBufferLatency,\n        });\n\n        rtpSource.pipe(jitterBuffer.input);\n        rtcpSource.pipe(time.input);\n\n        jitterBuffer.pipe(time.input);\n        time.pipe(depacketizer.input);\n        if (this.props.disableLipSync) {\n          depacketizer.pipe(webm.inputVideo);\n        } else {\n          depacketizer.pipe(lipsync.inputVideo);\n          lipsync.pipeVideo(webm.inputVideo);\n        }\n      } else {\n        const depacketizer = new DepacketizeCallback(codec);\n\n        rtpSource.pipe(time.input);\n        rtcpSource.pipe(time.input);\n\n        time.pipe(depacketizer.input);\n        if (this.props.disableLipSync) {\n          depacketizer.pipe(webm.inputAudio);\n        } else {\n          depacketizer.pipe(lipsync.inputAudio);\n          lipsync.pipeAudio(webm.inputAudio);\n        }\n      }\n\n      return rtpSource;\n    });\n    if (this.props.path) {\n      webm.pipe(saveToFileSystem(this.props.path));\n    } else if (this.props.stream) {\n      webm.pipe(async (o) => {\n        this.props.stream.execute(o);\n      });\n    }\n  }\n\n  async stop() {\n    await Promise.all(this.rtpSources.map((r) => r.stop()));\n    this.unSubscribers.dispose();\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = (typeof supportedVideoCodecs)[number];\n"]}
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{ unlink } from \"fs/promises\";\nimport { Event, EventDisposer } from \"../../../imports/common\";\n\nimport { MediaWriter } from \".\";\nimport { type MediaStreamTrack, WeriftError } from \"../../..\";\nimport {\n  DepacketizeCallback,\n  JitterBufferCallback,\n  LipsyncCallback,\n  NtpTimeCallback,\n  RtcpSourceCallback,\n  RtpSourceCallback,\n  RtpTimeCallback,\n  type SupportedCodec,\n  WebmCallback,\n  type WebmTrack,\n  saveToFileSystem,\n} from \"../../../../../rtp/src/extra\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n  rtpSources: RtpSourceCallback[] = [];\n  private onEol = new Event();\n  private ended = false;\n  unSubscribers = new EventDisposer();\n\n  async start(tracks: MediaStreamTrack[]) {\n    if (this.props.path) {\n      await unlink(this.props.path).catch((e) => e);\n    }\n\n    const inputTracks: (WebmTrack & { track: MediaStreamTrack })[] = tracks.map(\n      (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 WeriftError({\n                  message: \"unsupported codec\",\n                  payload: { track, path: sourcePath },\n                });\n            }\n          })();\n          return {\n            kind: \"video\" as const,\n            codec,\n            clockRate: 90000,\n            trackNumber,\n            width: this.props.width ?? 640,\n            height: this.props.height ?? 360,\n            roll: this.props.roll,\n            payloadType,\n            track,\n          };\n        } else {\n          return {\n            kind: \"audio\" as const,\n            codec: \"OPUS\" as const,\n            clockRate: 48000,\n            trackNumber,\n            payloadType,\n            track,\n          };\n        }\n      },\n    );\n\n    const webm = new WebmCallback(inputTracks, {\n      duration: this.props.defaultDuration ?? 1000 * 60 * 60 * 24,\n    });\n    const lipsync = new LipsyncCallback();\n\n    if (inputTracks.length === 1 || this.props.disableNtp) {\n      this.props.disableLipSync = true;\n    }\n\n    this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {\n      const rtpSource = new RtpSourceCallback();\n      const rtcpSource = new RtcpSourceCallback();\n      track.onReceiveRtp\n        .subscribe((rtp) => {\n          rtpSource.input(rtp.clone());\n        })\n        .disposer(this.unSubscribers);\n      track.onReceiveRtcp\n        .subscribe((rtcp) => {\n          rtcpSource.input(rtcp);\n        })\n        .disposer(this.unSubscribers);\n      const time = this.props.disableNtp\n        ? new RtpTimeCallback(clockRate)\n        : new NtpTimeCallback(clockRate);\n\n      if (track.kind === \"video\") {\n        const depacketizer = new DepacketizeCallback(codec, {\n          isFinalPacketInSequence: (h) => h.marker,\n        });\n        const jitterBuffer = new JitterBufferCallback(clockRate, {\n          bufferSize: this.props.jitterBufferSize,\n          latency: this.props.jitterBufferLatency,\n        });\n\n        rtpSource.pipe(jitterBuffer.input);\n        rtcpSource.pipe(time.input);\n\n        jitterBuffer.pipe(time.input);\n        time.pipe(depacketizer.input);\n        if (this.props.disableLipSync) {\n          depacketizer.pipe(webm.inputVideo);\n        } else {\n          depacketizer.pipe(lipsync.inputVideo);\n          lipsync.pipeVideo(webm.inputVideo);\n        }\n      } else {\n        const depacketizer = new DepacketizeCallback(codec);\n\n        rtpSource.pipe(time.input);\n        rtcpSource.pipe(time.input);\n\n        time.pipe(depacketizer.input);\n        if (this.props.disableLipSync) {\n          depacketizer.pipe(webm.inputAudio);\n        } else {\n          depacketizer.pipe(lipsync.inputAudio);\n          lipsync.pipeAudio(webm.inputAudio);\n        }\n      }\n\n      return rtpSource;\n    });\n    if (this.props.path) {\n      webm.pipe(async (o) => {\n        const eol = await saveToFileSystem(this.props.path)(o);\n        if (eol) {\n          this.onEol.execute();\n          this.ended = true;\n        }\n      });\n    } else if (this.props.stream) {\n      webm.pipe(async (o) => {\n        this.props.stream.execute(o);\n      });\n    }\n  }\n\n  async stop() {\n    await Promise.all(this.rtpSources.map((r) => r.stop()));\n\n    if (!this.ended) {\n      await this.onEol.asPromise(5000);\n    }\n    this.unSubscribers.dispose();\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = (typeof supportedVideoCodecs)[number];\n"]}

package/lib/webrtc/src/peerConnection.js

@@ -1198,8 +1198,9 @@ class RTCPeerConnection extends helper_1.EventTarget {
     addTrack(track, 
     /**todo impl */
     ms) {
-        if (this.isClosed)
+        if (this.isClosed) {
             throw new Error("is closed");
+        }
         if (this.getSenders().find((sender) => sender.track?.uuid === track.uuid)) {
             throw new Error("track exist");
         }

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

@@ -1 +1 @@
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debug from \"debug\";\nimport cloneDeep from \"lodash/cloneDeep\";\nimport isEqual from \"lodash/isEqual\";\nimport * as uuid from \"uuid\";\nimport { Event } from \"./imports/common\";\n\nimport {\n  type Address,\n  type CandidatePair,\n  type InterfaceAddresses,\n  Recvonly,\n  Sendonly,\n  Sendrecv,\n  deepMerge,\n} from \".\";\nimport {\n  type DtlsKeys,\n  codecParametersFromString,\n  useNACK,\n  usePLI,\n  useREMB,\n} from \".\";\nimport type { Profile } from \"../../dtls/src/context/srtp\";\nimport type { Message } from \"../../ice/src/stun/message\";\nimport type { Protocol } from \"../../ice/src/types/model\";\nimport {\n  DISCARD_HOST,\n  DISCARD_PORT,\n  ReceiverDirection,\n  SRTP_PROFILE,\n  SenderDirections,\n} from \"./const\";\nimport { RTCDataChannel, RTCDataChannelParameters } from \"./dataChannel\";\nimport { EventTarget, enumerate } from \"./helper\";\nimport {\n  RTCRtpCodecParameters,\n  RTCRtpCodingParameters,\n  type RTCRtpHeaderExtensionParameters,\n  type RTCRtpParameters,\n  type RTCRtpReceiveParameters,\n  RTCRtpRtxParameters,\n  RTCRtpSimulcastParameters,\n} from \"./media/parameters\";\nimport { RtpRouter } from \"./media/router\";\nimport { RTCRtpReceiver } from \"./media/rtpReceiver\";\nimport { RTCRtpSender } from \"./media/rtpSender\";\nimport {\n  type Direction,\n  RTCRtpTransceiver,\n  type TransceiverOptions,\n} from \"./media/rtpTransceiver\";\nimport { MediaStream, type MediaStreamTrack } from \"./media/track\";\nimport {\n  GroupDescription,\n  MediaDescription,\n  SessionDescription,\n  SsrcDescription,\n  addSDPHeader,\n} from \"./sdp\";\nimport { RTCCertificate, RTCDtlsTransport } from \"./transport/dtls\";\nimport {\n  IceCandidate,\n  type IceGathererState,\n  type RTCIceCandidate,\n  type RTCIceConnectionState,\n  RTCIceGatherer,\n  RTCIceTransport,\n} from \"./transport/ice\";\nimport { RTCSctpTransport } from \"./transport/sctp\";\nimport type { ConnectionState, Kind, RTCSignalingState } from \"./types/domain\";\nimport type { Callback, CallbackWithValue } from \"./types/util\";\nimport {\n  andDirection,\n  parseIceServers,\n  reverseDirection,\n  reverseSimulcastDirection,\n} from \"./utils\";\n\nconst log = debug(\"werift:packages/webrtc/src/peerConnection.ts\");\n\nexport class RTCPeerConnection extends EventTarget {\n  readonly cname = uuid.v4();\n  sctpTransport?: RTCSctpTransport;\n  config: Required<PeerConfig> = cloneDeep<PeerConfig>(defaultPeerConfig);\n  connectionState: ConnectionState = \"new\";\n  iceConnectionState: RTCIceConnectionState = \"new\";\n  iceGatheringState: IceGathererState = \"new\";\n  signalingState: RTCSignalingState = \"stable\";\n  negotiationneeded = false;\n  private readonly transceivers: RTCRtpTransceiver[] = [];\n  private pushTransceiver(t: RTCRtpTransceiver) {\n    this.transceivers.push(t);\n  }\n  private replaceTransceiver(t: RTCRtpTransceiver, index: number) {\n    this.transceivers[index] = t;\n  }\n  needRestart = false;\n\n  readonly iceGatheringStateChange = new Event<[IceGathererState]>();\n  readonly iceConnectionStateChange = new Event<[RTCIceConnectionState]>();\n  readonly signalingStateChange = new Event<[RTCSignalingState]>();\n  readonly connectionStateChange = new Event<[ConnectionState]>();\n  readonly onDataChannel = new Event<[RTCDataChannel]>();\n  readonly onRemoteTransceiverAdded = new Event<[RTCRtpTransceiver]>();\n  readonly onTransceiverAdded = new Event<[RTCRtpTransceiver]>();\n  readonly onIceCandidate = new Event<[RTCIceCandidate | undefined]>();\n  readonly onNegotiationneeded = new Event<[]>();\n  readonly onTrack = new Event<[MediaStreamTrack]>();\n\n  ondatachannel?: CallbackWithValue<RTCDataChannelEvent>;\n  onicecandidate?: CallbackWithValue<RTCPeerConnectionIceEvent>;\n  onnegotiationneeded?: CallbackWithValue<any>;\n  onsignalingstatechange?: CallbackWithValue<any>;\n  ontrack?: CallbackWithValue<RTCTrackEvent>;\n  onconnectionstatechange?: Callback;\n\n  private readonly router = new RtpRouter();\n  private certificate?: RTCCertificate;\n  sctpRemotePort?: number;\n  private seenMid = new Set<string>();\n  private currentLocalDescription?: SessionDescription;\n  private currentRemoteDescription?: SessionDescription;\n  private pendingLocalDescription?: SessionDescription;\n  private pendingRemoteDescription?: SessionDescription;\n  private isClosed = false;\n  private shouldNegotiationneeded = false;\n\n  get dtlsTransports() {\n    const transports = this.transceivers.map((t) => t.dtlsTransport);\n    if (this.sctpTransport) {\n      transports.push(this.sctpTransport.dtlsTransport);\n    }\n    return transports.reduce((acc: RTCDtlsTransport[], cur) => {\n      if (!acc.map((d) => d.id).includes(cur.id)) {\n        acc.push(cur);\n      }\n      return acc;\n    }, []);\n  }\n  get iceTransports() {\n    return this.dtlsTransports.map((d) => d.iceTransport);\n  }\n\n  get extIdUriMap() {\n    return this.router.extIdUriMap;\n  }\n\n  get iceGeneration() {\n    return this.iceTransports[0].connection.generation;\n  }\n\n  constructor(config: Partial<PeerConfig> = {}) {\n    super();\n\n    this.setConfiguration(config);\n\n    this.iceConnectionStateChange.subscribe((state) => {\n      switch (state) {\n        case \"disconnected\":\n          this.setConnectionState(\"disconnected\");\n          break;\n        case \"closed\":\n          this.close();\n          break;\n      }\n    });\n  }\n\n  setConfiguration(config: Partial<PeerConfig>) {\n    deepMerge(this.config, config);\n\n    if (this.config.icePortRange) {\n      const [min, max] = this.config.icePortRange;\n      if (min === max) throw new Error(\"should not be same value\");\n      if (min >= max) throw new Error(\"The min must be less than max\");\n    }\n\n    for (const [i, codecParams] of enumerate([\n      ...(this.config.codecs.audio || []),\n      ...(this.config.codecs.video || []),\n    ])) {\n      if (codecParams.payloadType != undefined) {\n        continue;\n      }\n\n      codecParams.payloadType = 96 + i;\n      switch (codecParams.name.toLowerCase()) {\n        case \"rtx\":\n          {\n            codecParams.parameters = `apt=${codecParams.payloadType - 1}`;\n          }\n          break;\n        case \"red\":\n          {\n            if (codecParams.contentType === \"audio\") {\n              const redundant = codecParams.payloadType + 1;\n              codecParams.parameters = `${redundant}/${redundant}`;\n              codecParams.payloadType = 63;\n            }\n          }\n          break;\n      }\n    }\n\n    [\n      ...(this.config.headerExtensions.audio || []),\n      ...(this.config.headerExtensions.video || []),\n    ].forEach((v, i) => {\n      v.id = 1 + i;\n    });\n\n    if (this.config.dtls) {\n      const { keys } = this.config.dtls;\n\n      if (keys) {\n        this.certificate = new RTCCertificate(\n          keys.keyPem,\n          keys.certPem,\n          keys.signatureHash,\n        );\n      }\n    }\n  }\n\n  get localDescription() {\n    if (!this._localDescription) {\n      return undefined;\n    }\n    return this._localDescription.toJSON();\n  }\n\n  get remoteDescription() {\n    if (!this._remoteDescription) {\n      return undefined;\n    }\n    return this._remoteDescription.toJSON();\n  }\n\n  /**@private */\n  get _localDescription() {\n    return this.pendingLocalDescription || this.currentLocalDescription;\n  }\n\n  /**@private */\n  get _remoteDescription() {\n    return this.pendingRemoteDescription || this.currentRemoteDescription;\n  }\n\n  private getTransceiverByMid(mid: string) {\n    return this.transceivers.find((transceiver) => transceiver.mid === mid);\n  }\n\n  private getTransceiverByMLineIndex(index: number) {\n    return this.transceivers.find(\n      (transceiver) => transceiver.mLineIndex === index,\n    );\n  }\n\n  getConfiguration() {\n    return this.config;\n  }\n\n  async createOffer({ iceRestart }: { iceRestart?: boolean } = {}) {\n    if (iceRestart || this.needRestart) {\n      this.needRestart = false;\n      for (const t of this.iceTransports) {\n        t.restart();\n      }\n    }\n\n    await this.ensureCerts();\n\n    const description = this.buildOfferSdp();\n    return description.toJSON();\n  }\n\n  private assignTransceiverCodecs(transceiver: RTCRtpTransceiver) {\n    const codecs = (\n      this.config.codecs[transceiver.kind] as RTCRtpCodecParameters[]\n    ).filter((codecCandidate) => {\n      switch (codecCandidate.direction) {\n        case \"recvonly\": {\n          if (ReceiverDirection.includes(transceiver.direction)) return true;\n          return false;\n        }\n        case \"sendonly\": {\n          if (SenderDirections.includes(transceiver.direction)) return true;\n          return false;\n        }\n        case \"sendrecv\": {\n          if ([Sendrecv, Recvonly, Sendonly].includes(transceiver.direction))\n            return true;\n          return false;\n        }\n        case \"all\": {\n          return true;\n        }\n        default:\n          return false;\n      }\n    });\n    transceiver.codecs = codecs;\n  }\n\n  buildOfferSdp() {\n    this.transceivers.forEach((transceiver) => {\n      if (transceiver.codecs.length === 0) {\n        this.assignTransceiverCodecs(transceiver);\n      }\n      if (transceiver.headerExtensions.length === 0) {\n        transceiver.headerExtensions =\n          this.config.headerExtensions[transceiver.kind] ?? [];\n      }\n    });\n\n    const description = new SessionDescription();\n    addSDPHeader(\"offer\", description);\n\n    // # handle existing transceivers / sctp\n\n    const currentMedia = this._localDescription\n      ? this._localDescription.media\n      : [];\n\n    currentMedia.forEach((m, i) => {\n      const mid = m.rtp.muxId;\n      if (!mid) {\n        log(\"mid missing\", m);\n        return;\n      }\n      if (m.kind === \"application\") {\n        if (!this.sctpTransport) {\n          throw new Error(\"sctpTransport not found\");\n        }\n        this.sctpTransport.mLineIndex = i;\n        description.media.push(\n          createMediaDescriptionForSctp(this.sctpTransport),\n        );\n      } else {\n        const transceiver = this.getTransceiverByMid(mid);\n        if (!transceiver) {\n          if (m.direction === \"inactive\") {\n            description.media.push(m);\n            return;\n          }\n          throw new Error(\"transceiver not found\");\n        }\n        transceiver.mLineIndex = i;\n        description.media.push(\n          createMediaDescriptionForTransceiver(\n            transceiver,\n            this.cname,\n            transceiver.direction,\n          ),\n        );\n      }\n    });\n\n    // # handle new transceivers / sctp\n    this.transceivers\n      .filter((t) => !description.media.find((m) => m.rtp.muxId === t.mid))\n      .forEach((transceiver) => {\n        if (transceiver.mid == undefined) {\n          transceiver.mid = allocateMid(\n            this.seenMid,\n            this.config.midSuffix ? \"av\" : \"\",\n          );\n        }\n        const mediaDescription = createMediaDescriptionForTransceiver(\n          transceiver,\n          this.cname,\n          transceiver.direction,\n        );\n        if (transceiver.mLineIndex === undefined) {\n          transceiver.mLineIndex = description.media.length;\n          description.media.push(mediaDescription);\n        } else {\n          description.media[transceiver.mLineIndex] = mediaDescription;\n        }\n      });\n\n    if (\n      this.sctpTransport &&\n      !description.media.find((m) => m.kind === \"application\")\n    ) {\n      this.sctpTransport.mLineIndex = description.media.length;\n      if (this.sctpTransport.mid == undefined) {\n        this.sctpTransport.mid = allocateMid(\n          this.seenMid,\n          this.config.midSuffix ? \"dc\" : \"\",\n        );\n      }\n      description.media.push(createMediaDescriptionForSctp(this.sctpTransport));\n    }\n\n    if (this.config.bundlePolicy !== \"disable\") {\n      const mids = description.media\n        .map((m) => (m.direction !== \"inactive\" ? m.rtp.muxId : undefined))\n        .filter((v) => v) as string[];\n      if (mids.length) {\n        const bundle = new GroupDescription(\"BUNDLE\", mids);\n        description.group.push(bundle);\n      }\n    }\n\n    return description;\n  }\n\n  createDataChannel(\n    label: string,\n    options: Partial<{\n      maxPacketLifeTime?: number;\n      protocol: string;\n      maxRetransmits?: number;\n      ordered: boolean;\n      negotiated: boolean;\n      id?: number;\n    }> = {},\n  ): RTCDataChannel {\n    const base: typeof options = {\n      protocol: \"\",\n      ordered: true,\n      negotiated: false,\n    };\n    const settings: Required<typeof base> = { ...base, ...options } as any;\n\n    if (settings.maxPacketLifeTime && settings.maxRetransmits) {\n      throw new Error(\"can not select both\");\n    }\n\n    if (!this.sctpTransport) {\n      this.sctpTransport = this.createSctpTransport();\n      this.needNegotiation();\n    }\n\n    const parameters = new RTCDataChannelParameters({\n      id: settings.id,\n      label,\n      maxPacketLifeTime: settings.maxPacketLifeTime,\n      maxRetransmits: settings.maxRetransmits,\n      negotiated: settings.negotiated,\n      ordered: settings.ordered,\n      protocol: settings.protocol,\n    });\n\n    const channel = new RTCDataChannel(this.sctpTransport, parameters);\n    return channel;\n  }\n\n  removeTrack(sender: RTCRtpSender) {\n    if (this.isClosed) throw new Error(\"peer closed\");\n    if (!this.getSenders().find(({ ssrc }) => sender.ssrc === ssrc)) {\n      throw new Error(\"unExist\");\n    }\n\n    const transceiver = this.transceivers.find(\n      ({ sender: { ssrc } }) => sender.ssrc === ssrc,\n    );\n    if (!transceiver) throw new Error(\"unExist\");\n\n    sender.stop();\n\n    if (transceiver.currentDirection === \"recvonly\") {\n      this.needNegotiation();\n      return;\n    }\n\n    if (transceiver.stopping || transceiver.stopped) {\n      transceiver.setDirection(\"inactive\");\n    } else {\n      if (transceiver.direction === \"sendrecv\") {\n        transceiver.setDirection(\"recvonly\");\n      } else if (\n        transceiver.direction === \"sendonly\" ||\n        transceiver.direction === \"recvonly\"\n      ) {\n        transceiver.setDirection(\"inactive\");\n      }\n    }\n    this.needNegotiation();\n  }\n\n  private needNegotiation = async () => {\n    this.shouldNegotiationneeded = true;\n    if (this.negotiationneeded || this.signalingState !== \"stable\") {\n      return;\n    }\n    this.shouldNegotiationneeded = false;\n    setImmediate(() => {\n      this.negotiationneeded = true;\n      this.onNegotiationneeded.execute();\n      if (this.onnegotiationneeded) this.onnegotiationneeded({});\n    });\n  };\n\n  private createTransport(srtpProfiles: Profile[] = []) {\n    const [existing] = this.iceTransports;\n\n    // Gather ICE candidates for only one track. If the remote endpoint is not bundle-aware, negotiate only one media track.\n    // https://w3c.github.io/webrtc-pc/#rtcbundlepolicy-enum\n    if (this.config.bundlePolicy === \"max-bundle\") {\n      if (existing) {\n        return this.dtlsTransports[0];\n      }\n    }\n\n    const iceGatherer = new RTCIceGatherer({\n      ...parseIceServers(this.config.iceServers),\n      forceTurn: this.config.iceTransportPolicy === \"relay\",\n      portRange: this.config.icePortRange,\n      interfaceAddresses: this.config.iceInterfaceAddresses,\n      additionalHostAddresses: this.config.iceAdditionalHostAddresses,\n      filterStunResponse: this.config.iceFilterStunResponse,\n      filterCandidatePair: this.config.iceFilterCandidatePair,\n      localPasswordPrefix: this.config.icePasswordPrefix,\n      useIpv4: this.config.iceUseIpv4,\n      useIpv6: this.config.iceUseIpv6,\n      turnTransport: this.config.forceTurnTCP === true ? \"tcp\" : \"udp\",\n    });\n    if (existing) {\n      iceGatherer.connection.localUsername = existing.connection.localUsername;\n      iceGatherer.connection.localPassword = existing.connection.localPassword;\n    }\n    iceGatherer.onGatheringStateChange.subscribe(() => {\n      this.updateIceGatheringState();\n    });\n    this.updateIceGatheringState();\n    const iceTransport = new RTCIceTransport(iceGatherer);\n    iceTransport.onStateChange.subscribe(() => {\n      this.updateIceConnectionState();\n    });\n    iceTransport.onNegotiationNeeded.subscribe(() => {\n      this.needNegotiation();\n    });\n    iceTransport.onIceCandidate.subscribe((candidate) => {\n      if (!this.localDescription) {\n        log(\"localDescription not found when ice candidate was gathered\");\n        return;\n      }\n      if (!candidate) {\n        this.setLocal(this._localDescription!);\n        this.onIceCandidate.execute(undefined);\n        if (this.onicecandidate) {\n          this.onicecandidate({ candidate: undefined });\n        }\n        this.emit(\"icecandidate\", { candidate: undefined });\n        return;\n      }\n\n      if (this.config.bundlePolicy === \"max-bundle\" || this.remoteIsBundled) {\n        candidate.sdpMLineIndex = 0;\n        const media = this._localDescription?.media[0];\n        if (media) {\n          candidate.sdpMid = media.rtp.muxId;\n        }\n      } else {\n        const transceiver = this.transceivers.find(\n          (t) => t.dtlsTransport.iceTransport.id === iceTransport.id,\n        );\n        if (transceiver) {\n          candidate.sdpMLineIndex = transceiver.mLineIndex;\n          candidate.sdpMid = transceiver.mid;\n        }\n        if (\n          this.sctpTransport?.dtlsTransport.iceTransport.id === iceTransport.id\n        ) {\n          candidate.sdpMLineIndex = this.sctpTransport.mLineIndex;\n          candidate.sdpMid = this.sctpTransport.mid;\n        }\n      }\n\n      candidate.foundation = \"candidate:\" + candidate.foundation;\n\n      this.onIceCandidate.execute(candidate.toJSON());\n      if (this.onicecandidate) {\n        this.onicecandidate({ candidate: candidate.toJSON() });\n      }\n      this.emit(\"icecandidate\", { candidate });\n    });\n\n    const dtlsTransport = new RTCDtlsTransport(\n      this.config,\n      iceTransport,\n      this.router,\n      this.certificate,\n      srtpProfiles,\n    );\n\n    return dtlsTransport;\n  }\n\n  private createSctpTransport() {\n    const dtlsTransport = this.createTransport([\n      SRTP_PROFILE.SRTP_AEAD_AES_128_GCM, // prefer\n      SRTP_PROFILE.SRTP_AES128_CM_HMAC_SHA1_80,\n    ]);\n    const sctp = new RTCSctpTransport();\n    sctp.setDtlsTransport(dtlsTransport);\n    sctp.mid = undefined;\n\n    sctp.onDataChannel.subscribe((channel) => {\n      this.onDataChannel.execute(channel);\n\n      const event: RTCDataChannelEvent = { channel };\n      if (this.ondatachannel) this.ondatachannel(event);\n      this.emit(\"datachannel\", event);\n    });\n\n    this.sctpTransport = sctp;\n    this.updateIceConnectionState();\n\n    return sctp;\n  }\n\n  async setLocalDescription(sessionDescription: {\n    type: \"offer\" | \"answer\";\n    sdp: string;\n  }): Promise<SessionDescription> {\n    // # parse and validate description\n    const description = SessionDescription.parse(sessionDescription.sdp);\n    description.type = sessionDescription.type;\n    this.validateDescription(description, true);\n\n    // # update signaling state\n    if (description.type === \"offer\") {\n      this.setSignalingState(\"have-local-offer\");\n    } else if (description.type === \"answer\") {\n      this.setSignalingState(\"stable\");\n    }\n\n    // # assign MID\n    for (const [i, media] of enumerate(description.media)) {\n      const mid = media.rtp.muxId!;\n      this.seenMid.add(mid);\n      if ([\"audio\", \"video\"].includes(media.kind)) {\n        const transceiver = this.getTransceiverByMLineIndex(i);\n        if (transceiver) {\n          transceiver.mid = mid;\n        }\n      }\n      if (media.kind === \"application\" && this.sctpTransport) {\n        this.sctpTransport.mid = mid;\n      }\n    }\n\n    // setup ice,dtls role\n    for (const dtlsTransport of this.dtlsTransports) {\n      const iceTransport = dtlsTransport.iceTransport;\n\n      // # set ICE role\n      if (description.type === \"offer\") {\n        iceTransport.connection.iceControlling = true;\n      } else {\n        iceTransport.connection.iceControlling = false;\n      }\n      // One agent full, one lite:  The full agent MUST take the controlling role, and the lite agent MUST take the controlled role\n      // RFC 8445 S6.1.1\n      if (iceTransport.connection.remoteIsLite) {\n        iceTransport.connection.iceControlling = true;\n      }\n\n      // # set DTLS role for mediasoup\n      if (description.type === \"answer\") {\n        const role = description.media.find((media) => media.dtlsParams)\n          ?.dtlsParams?.role;\n        if (role) {\n          dtlsTransport.role = role;\n        }\n      }\n    }\n\n    // # configure direction\n    if ([\"answer\", \"pranswer\"].includes(description.type)) {\n      for (const t of this.transceivers) {\n        const direction = andDirection(t.direction, t.offerDirection);\n        t.setCurrentDirection(direction);\n      }\n    }\n\n    // for trickle ice\n    this.setLocal(description);\n\n    await this.gatherCandidates().catch((e) => {\n      log(\"gatherCandidates failed\", e);\n    });\n\n    // connect transports\n    if (description.type === \"answer\") {\n      this.connect().catch((err) => {\n        log(\"connect failed\", err);\n        this.setConnectionState(\"failed\");\n      });\n    }\n\n    this.setLocal(description);\n\n    if (this.shouldNegotiationneeded) {\n      this.needNegotiation();\n    }\n\n    return description;\n  }\n\n  private async gatherCandidates() {\n    // # gather candidates\n    const connected = this.iceTransports.find(\n      (transport) => transport.state === \"connected\",\n    );\n    if (this.remoteIsBundled && connected) {\n      // no need to gather ice candidates on an existing bundled connection\n    } else {\n      await Promise.allSettled(\n        this.iceTransports.map((iceTransport) => iceTransport.gather()),\n      );\n    }\n  }\n\n  private setLocal(description: SessionDescription) {\n    description.media\n      .filter((m) => [\"audio\", \"video\"].includes(m.kind))\n      .forEach((m, i) => {\n        addTransportDescription(m, this.transceivers[i].dtlsTransport);\n      });\n    const sctpMedia = description.media.find((m) => m.kind === \"application\");\n    if (this.sctpTransport && sctpMedia) {\n      addTransportDescription(sctpMedia, this.sctpTransport.dtlsTransport);\n    }\n\n    if (description.type === \"answer\") {\n      this.currentLocalDescription = description;\n      this.pendingLocalDescription = undefined;\n    } else {\n      this.pendingLocalDescription = description;\n    }\n  }\n\n  private getTransportByMid(mid: string) {\n    let iceTransport: RTCIceTransport | undefined;\n\n    const transceiver = this.transceivers.find((t) => t.mid === mid);\n    if (transceiver) {\n      iceTransport = transceiver.dtlsTransport.iceTransport;\n    } else if (!iceTransport && this.sctpTransport?.mid === mid) {\n      iceTransport = this.sctpTransport?.dtlsTransport.iceTransport;\n    }\n\n    return iceTransport;\n  }\n\n  private getTransportByMLineIndex(index: number) {\n    const sdp = this.buildOfferSdp();\n    const media = sdp.media[index];\n    if (!media) {\n      return;\n    }\n    const transport = this.getTransportByMid(media.rtp.muxId!);\n\n    return transport;\n  }\n\n  async addIceCandidate(candidateMessage: RTCIceCandidate) {\n    const candidate = IceCandidate.fromJSON(candidateMessage);\n    if (!candidate) {\n      return;\n    }\n\n    let iceTransport: RTCIceTransport | undefined;\n\n    if (typeof candidate.sdpMid === \"number\") {\n      iceTransport = this.getTransportByMid(candidate.sdpMid);\n    }\n\n    if (!iceTransport && typeof candidate.sdpMLineIndex === \"number\") {\n      iceTransport = this.getTransportByMLineIndex(candidate.sdpMLineIndex);\n    }\n\n    if (!iceTransport) {\n      iceTransport = this.iceTransports[0];\n    }\n\n    if (iceTransport) {\n      await iceTransport.addRemoteCandidate(candidate);\n    } else {\n      log(\"iceTransport not found\", candidate);\n    }\n  }\n\n  private async connect() {\n    log(\"start connect\");\n\n    await Promise.allSettled(\n      this.dtlsTransports.map(async (dtlsTransport) => {\n        const { iceTransport } = dtlsTransport;\n        if (iceTransport.state === \"connected\") {\n          return;\n        }\n\n        this.setConnectionState(\"connecting\");\n\n        await iceTransport.start().catch((err) => {\n          log(\"iceTransport.start failed\", err);\n          throw err;\n        });\n\n        if (dtlsTransport.state === \"connected\") {\n          return;\n        }\n        await dtlsTransport.start().catch((err) => {\n          log(\"dtlsTransport.start failed\", err);\n          throw err;\n        });\n        if (\n          this.sctpTransport &&\n          this.sctpRemotePort &&\n          this.sctpTransport.dtlsTransport.id === dtlsTransport.id\n        ) {\n          await this.sctpTransport.start(this.sctpRemotePort);\n          await this.sctpTransport.sctp.stateChanged.connected.asPromise();\n          log(\"sctp connected\");\n        }\n      }),\n    );\n\n    this.setConnectionState(\"connected\");\n  }\n\n  private getLocalRtpParams(transceiver: RTCRtpTransceiver): RTCRtpParameters {\n    if (transceiver.mid == undefined) throw new Error(\"mid not assigned\");\n\n    const rtp: RTCRtpParameters = {\n      codecs: transceiver.codecs,\n      muxId: transceiver.mid,\n      headerExtensions: transceiver.headerExtensions,\n      rtcp: { cname: this.cname, ssrc: transceiver.sender.ssrc, mux: true },\n    };\n    return rtp;\n  }\n\n  private getRemoteRtpParams(\n    media: MediaDescription,\n    transceiver: RTCRtpTransceiver,\n  ): RTCRtpReceiveParameters {\n    const receiveParameters: RTCRtpReceiveParameters = {\n      muxId: media.rtp.muxId,\n      rtcp: media.rtp.rtcp,\n      codecs: transceiver.codecs,\n      headerExtensions: transceiver.headerExtensions,\n      encodings: Object.values(\n        transceiver.codecs.reduce(\n          (acc: { [pt: number]: RTCRtpCodingParameters }, codec) => {\n            if (codec.name.toLowerCase() === \"rtx\") {\n              const params = codecParametersFromString(codec.parameters ?? \"\");\n              const apt = acc[params[\"apt\"]];\n              if (apt && media.ssrc.length === 2) {\n                apt.rtx = new RTCRtpRtxParameters({ ssrc: media.ssrc[1].ssrc });\n              }\n              return acc;\n            }\n            acc[codec.payloadType] = new RTCRtpCodingParameters({\n              ssrc: media.ssrc[0]?.ssrc,\n              payloadType: codec.payloadType,\n            });\n            return acc;\n          },\n          {},\n        ),\n      ),\n    };\n\n    return receiveParameters;\n  }\n\n  private get remoteIsBundled() {\n    const remoteSdp = this._remoteDescription;\n    if (!remoteSdp) {\n      return undefined;\n    }\n    const bundle = remoteSdp.group.find(\n      (g) => g.semantic === \"BUNDLE\" && this.config.bundlePolicy !== \"disable\",\n    );\n    return bundle;\n  }\n\n  restartIce() {\n    this.needRestart = true;\n    this.needNegotiation();\n  }\n\n  async setRemoteDescription(sessionDescription: RTCSessionDescriptionInit) {\n    if (\n      !sessionDescription.sdp ||\n      !sessionDescription.type ||\n      sessionDescription.type === \"rollback\" ||\n      sessionDescription.type === \"pranswer\"\n    ) {\n      throw new Error(\"invalid sessionDescription\");\n    }\n\n    // # parse and validate description\n    const remoteSdp = SessionDescription.parse(sessionDescription.sdp);\n    remoteSdp.type = sessionDescription.type;\n    this.validateDescription(remoteSdp, false);\n\n    if (remoteSdp.type === \"answer\") {\n      this.currentRemoteDescription = remoteSdp;\n      this.pendingRemoteDescription = undefined;\n    } else {\n      this.pendingRemoteDescription = remoteSdp;\n    }\n\n    let bundleTransport: RTCDtlsTransport | undefined;\n\n    // # apply description\n\n    const matchTransceiverWithMedia = (\n      transceiver: RTCRtpTransceiver,\n      media: MediaDescription,\n    ) =>\n      transceiver.kind === media.kind &&\n      [undefined, media.rtp.muxId].includes(transceiver.mid);\n\n    let transports = remoteSdp.media.map((remoteMedia, i) => {\n      let dtlsTransport: RTCDtlsTransport;\n\n      if ([\"audio\", \"video\"].includes(remoteMedia.kind)) {\n        let transceiver = this.transceivers.find((t) =>\n          matchTransceiverWithMedia(t, remoteMedia),\n        );\n        if (!transceiver) {\n          // create remote transceiver\n          transceiver = this._addTransceiver(remoteMedia.kind, {\n            direction: \"recvonly\",\n          });\n          transceiver.mid = remoteMedia.rtp.muxId;\n          this.onRemoteTransceiverAdded.execute(transceiver);\n        } else {\n          if (transceiver.direction === \"inactive\" && transceiver.stopping) {\n            transceiver.stopped = true;\n\n            if (sessionDescription.type === \"answer\") {\n              transceiver.setCurrentDirection(\"inactive\");\n            }\n            return;\n          }\n        }\n\n        if (this.remoteIsBundled) {\n          if (!bundleTransport) {\n            bundleTransport = transceiver.dtlsTransport;\n          } else {\n            transceiver.setDtlsTransport(bundleTransport);\n          }\n        }\n\n        dtlsTransport = transceiver.dtlsTransport;\n\n        this.setRemoteRTP(transceiver, remoteMedia, remoteSdp.type, i);\n      } else if (remoteMedia.kind === \"application\") {\n        if (!this.sctpTransport) {\n          this.sctpTransport = this.createSctpTransport();\n          this.sctpTransport.mid = remoteMedia.rtp.muxId;\n        }\n\n        if (this.remoteIsBundled) {\n          if (!bundleTransport) {\n            bundleTransport = this.sctpTransport.dtlsTransport;\n          } else {\n            this.sctpTransport.setDtlsTransport(bundleTransport);\n          }\n        }\n\n        dtlsTransport = this.sctpTransport.dtlsTransport;\n\n        this.setRemoteSCTP(remoteMedia, this.sctpTransport, i);\n      } else {\n        throw new Error(\"invalid media kind\");\n      }\n\n      const iceTransport = dtlsTransport.iceTransport;\n\n      if (remoteMedia.iceParams) {\n        const renomination = !!remoteSdp.media.find(\n          (m) => m.direction === \"inactive\",\n        );\n        iceTransport.setRemoteParams(remoteMedia.iceParams, renomination);\n\n        // One agent full, one lite:  The full agent MUST take the controlling role, and the lite agent MUST take the controlled role\n        // RFC 8445 S6.1.1\n        if (remoteMedia.iceParams?.iceLite) {\n          iceTransport.connection.iceControlling = true;\n        }\n      }\n      if (remoteMedia.dtlsParams) {\n        dtlsTransport.setRemoteParams(remoteMedia.dtlsParams);\n      }\n\n      // # add ICE candidates\n      remoteMedia.iceCandidates.forEach(iceTransport.addRemoteCandidate);\n\n      if (remoteMedia.iceCandidatesComplete) {\n        iceTransport.addRemoteCandidate(undefined);\n      }\n\n      // # set DTLS role\n      if (remoteSdp.type === \"answer\" && remoteMedia.dtlsParams?.role) {\n        dtlsTransport.role =\n          remoteMedia.dtlsParams.role === \"client\" ? \"server\" : \"client\";\n      }\n      return iceTransport;\n    }) as RTCIceTransport[];\n\n    // filter out inactive transports\n    transports = transports.filter((iceTransport) => !!iceTransport);\n\n    const removedTransceivers = this.transceivers.filter(\n      (t) =>\n        remoteSdp.media.find((m) => matchTransceiverWithMedia(t, m)) ==\n        undefined,\n    );\n\n    if (sessionDescription.type === \"answer\") {\n      for (const transceiver of removedTransceivers) {\n        // todo: handle answer side transceiver removal work.\n        // event should trigger to notify media source to stop.\n        transceiver.stop();\n        transceiver.stopped = true;\n      }\n    }\n\n    if (remoteSdp.type === \"offer\") {\n      this.setSignalingState(\"have-remote-offer\");\n    } else if (remoteSdp.type === \"answer\") {\n      this.setSignalingState(\"stable\");\n    }\n\n    // connect transports\n    if (remoteSdp.type === \"answer\") {\n      log(\"caller start connect\");\n      this.connect().catch((err) => {\n        log(\"connect failed\", err);\n        this.setConnectionState(\"failed\");\n      });\n    }\n\n    this.negotiationneeded = false;\n    if (this.shouldNegotiationneeded) {\n      this.needNegotiation();\n    }\n  }\n\n  private setRemoteRTP(\n    transceiver: RTCRtpTransceiver,\n    remoteMedia: MediaDescription,\n    type: \"offer\" | \"answer\",\n    mLineIndex: number,\n  ) {\n    if (!transceiver.mid) {\n      transceiver.mid = remoteMedia.rtp.muxId;\n    }\n    transceiver.mLineIndex = mLineIndex;\n\n    // # negotiate codecs\n    transceiver.codecs = remoteMedia.rtp.codecs.filter((remoteCodec) => {\n      const localCodecs = this.config.codecs[remoteMedia.kind] || [];\n\n      const existCodec = findCodecByMimeType(localCodecs, remoteCodec);\n      if (!existCodec) {\n        return false;\n      }\n\n      if (existCodec?.name.toLowerCase() === \"rtx\") {\n        const params = codecParametersFromString(existCodec.parameters ?? \"\");\n        const pt = params[\"apt\"];\n        const origin = remoteMedia.rtp.codecs.find((c) => c.payloadType === pt);\n        if (!origin) {\n          return false;\n        }\n        return !!findCodecByMimeType(localCodecs, origin);\n      }\n\n      return true;\n    });\n\n    log(\"negotiated codecs\", transceiver.codecs);\n    if (transceiver.codecs.length === 0) {\n      throw new Error(\"negotiate codecs failed.\");\n    }\n    transceiver.headerExtensions = remoteMedia.rtp.headerExtensions.filter(\n      (extension) =>\n        (this.config.headerExtensions[remoteMedia.kind as Media] || []).find(\n          (v) => v.uri === extension.uri,\n        ),\n    );\n\n    // # configure direction\n    const mediaDirection = remoteMedia.direction ?? \"inactive\";\n    const direction = reverseDirection(mediaDirection);\n    if ([\"answer\", \"pranswer\"].includes(type)) {\n      transceiver.setCurrentDirection(direction);\n    } else {\n      transceiver.offerDirection = direction;\n    }\n    const localParams = this.getLocalRtpParams(transceiver);\n    transceiver.sender.prepareSend(localParams);\n\n    if ([\"recvonly\", \"sendrecv\"].includes(transceiver.direction)) {\n      const remotePrams = this.getRemoteRtpParams(remoteMedia, transceiver);\n\n      // register simulcast receiver\n      remoteMedia.simulcastParameters.forEach((param) => {\n        this.router.registerRtpReceiverByRid(transceiver, param, remotePrams);\n      });\n\n      transceiver.receiver.prepareReceive(remotePrams);\n      // register ssrc receiver\n      this.router.registerRtpReceiverBySsrc(transceiver, remotePrams);\n    }\n    if ([\"sendonly\", \"sendrecv\"].includes(mediaDirection)) {\n      // assign msid\n      if (remoteMedia.msid != undefined) {\n        const [streamId, trackId] = remoteMedia.msid.split(\" \");\n        transceiver.receiver.remoteStreamId = streamId;\n        transceiver.receiver.remoteTrackId = trackId;\n\n        this.fireOnTrack(\n          transceiver.receiver.track,\n          transceiver,\n          new MediaStream({\n            id: streamId,\n            tracks: [transceiver.receiver.track],\n          }),\n        );\n      }\n    }\n\n    if (remoteMedia.ssrc[0]?.ssrc) {\n      transceiver.receiver.setupTWCC(remoteMedia.ssrc[0].ssrc);\n    }\n  }\n\n  private setRemoteSCTP(\n    remoteMedia: MediaDescription,\n    sctpTransport: RTCSctpTransport,\n    mLineIndex: number,\n  ) {\n    // # configure sctp\n    this.sctpRemotePort = remoteMedia.sctpPort;\n    if (!this.sctpRemotePort) {\n      throw new Error(\"sctpRemotePort not exist\");\n    }\n\n    sctpTransport.setRemotePort(this.sctpRemotePort);\n    sctpTransport.mLineIndex = mLineIndex;\n    if (!sctpTransport.mid) {\n      sctpTransport.mid = remoteMedia.rtp.muxId;\n    }\n  }\n\n  private validateDescription(\n    description: SessionDescription,\n    isLocal: boolean,\n  ) {\n    if (isLocal) {\n      if (description.type === \"offer\") {\n        if (![\"stable\", \"have-local-offer\"].includes(this.signalingState))\n          throw new Error(\"Cannot handle offer in signaling state\");\n      } else if (description.type === \"answer\") {\n        if (\n          ![\"have-remote-offer\", \"have-local-pranswer\"].includes(\n            this.signalingState,\n          )\n        ) {\n          throw new Error(\"Cannot handle answer in signaling state\");\n        }\n      }\n    } else {\n      if (description.type === \"offer\") {\n        if (![\"stable\", \"have-remote-offer\"].includes(this.signalingState)) {\n          throw new Error(\"Cannot handle offer in signaling state\");\n        }\n      } else if (description.type === \"answer\") {\n        if (\n          ![\"have-local-offer\", \"have-remote-pranswer\"].includes(\n            this.signalingState,\n          )\n        ) {\n          throw new Error(\"Cannot handle answer in signaling state\");\n        }\n      }\n    }\n\n    description.media.forEach((media) => {\n      if (media.direction === \"inactive\") return;\n      if (\n        !media.iceParams ||\n        !media.iceParams.usernameFragment ||\n        !media.iceParams.password\n      )\n        throw new Error(\"ICE username fragment or password is missing\");\n    });\n\n    if ([\"answer\", \"pranswer\"].includes(description.type || \"\")) {\n      const offer = isLocal ? this._remoteDescription : this._localDescription;\n      if (!offer) throw new Error();\n\n      const answerMedia = description.media.map((v, i) => [v.kind, i]);\n      const offerMedia = offer.media.map((v, i) => [v.kind, i]);\n      if (!isEqual(offerMedia, answerMedia)) {\n        throw new Error(\"Media sections in answer do not match offer\");\n      }\n    }\n  }\n\n  private fireOnTrack(\n    track: MediaStreamTrack,\n    transceiver: RTCRtpTransceiver,\n    stream: MediaStream,\n  ) {\n    const event: RTCTrackEvent = {\n      track,\n      streams: [stream],\n      transceiver,\n      receiver: transceiver.receiver,\n    };\n    this.onTrack.execute(track);\n    this.emit(\"track\", event);\n    if (this.ontrack) this.ontrack(event);\n  }\n\n  addTransceiver(\n    trackOrKind: Kind | MediaStreamTrack,\n    options: Partial<TransceiverOptions> = {},\n  ) {\n    return this._addTransceiver(trackOrKind, options);\n  }\n\n  private _addTransceiver(\n    trackOrKind: Kind | MediaStreamTrack,\n    options: Partial<TransceiverOptions> = {},\n  ) {\n    const kind =\n      typeof trackOrKind === \"string\" ? trackOrKind : trackOrKind.kind;\n\n    const direction = options.direction || \"sendrecv\";\n\n    const dtlsTransport = this.createTransport([\n      SRTP_PROFILE.SRTP_AEAD_AES_128_GCM, // prefer\n      SRTP_PROFILE.SRTP_AES128_CM_HMAC_SHA1_80,\n    ]);\n\n    const sender = new RTCRtpSender(trackOrKind);\n    const receiver = new RTCRtpReceiver(this.config, kind, sender.ssrc);\n    const newTransceiver = new RTCRtpTransceiver(\n      kind,\n      dtlsTransport,\n      receiver,\n      sender,\n      direction,\n    );\n    newTransceiver.options = options;\n    this.router.registerRtpSender(newTransceiver.sender);\n\n    // reuse inactive\n    const inactiveTransceiverIndex = this.transceivers.findIndex(\n      (t) => t.currentDirection === \"inactive\",\n    );\n    const inactiveTransceiver = this.transceivers.find(\n      (t) => t.currentDirection === \"inactive\",\n    );\n    if (inactiveTransceiverIndex > -1 && inactiveTransceiver) {\n      this.replaceTransceiver(newTransceiver, inactiveTransceiverIndex);\n      newTransceiver.mLineIndex = inactiveTransceiver.mLineIndex;\n      inactiveTransceiver.setCurrentDirection(undefined);\n    } else {\n      this.pushTransceiver(newTransceiver);\n    }\n    this.onTransceiverAdded.execute(newTransceiver);\n\n    this.updateIceConnectionState();\n    this.needNegotiation();\n\n    return newTransceiver;\n  }\n\n  getTransceivers() {\n    return this.transceivers;\n  }\n\n  getSenders(): RTCRtpSender[] {\n    return this.getTransceivers().map((t) => t.sender);\n  }\n\n  getReceivers() {\n    return this.getTransceivers().map((t) => t.receiver);\n  }\n\n  // todo fix\n  addTrack(\n    track: MediaStreamTrack,\n    /**todo impl */\n    ms?: MediaStream,\n  ) {\n    if (this.isClosed) throw new Error(\"is closed\");\n    if (this.getSenders().find((sender) => sender.track?.uuid === track.uuid)) {\n      throw new Error(\"track exist\");\n    }\n\n    const emptyTrackSender = this.transceivers.find(\n      (t) =>\n        t.sender.track == undefined &&\n        t.kind === track.kind &&\n        SenderDirections.includes(t.direction) === true,\n    );\n    if (emptyTrackSender) {\n      const sender = emptyTrackSender.sender;\n      sender.registerTrack(track);\n      this.needNegotiation();\n      return sender;\n    }\n\n    const notSendTransceiver = this.transceivers.find(\n      (t) =>\n        t.sender.track == undefined &&\n        t.kind === track.kind &&\n        SenderDirections.includes(t.direction) === false &&\n        !t.usedForSender,\n    );\n    if (notSendTransceiver) {\n      const sender = notSendTransceiver.sender;\n      sender.registerTrack(track);\n      switch (notSendTransceiver.direction) {\n        case \"recvonly\":\n          notSendTransceiver.setDirection(\"sendrecv\");\n          break;\n        case \"inactive\":\n          notSendTransceiver.setDirection(\"sendonly\");\n          break;\n      }\n      this.needNegotiation();\n      return sender;\n    } else {\n      const transceiver = this._addTransceiver(track, {\n        direction: \"sendrecv\",\n      });\n      this.needNegotiation();\n      return transceiver.sender;\n    }\n  }\n\n  private async ensureCerts() {\n    if (!this.certificate) {\n      this.certificate = await RTCDtlsTransport.SetupCertificate();\n    }\n\n    for (const dtlsTransport of this.dtlsTransports) {\n      dtlsTransport.localCertificate = this.certificate;\n    }\n  }\n\n  async createAnswer() {\n    await this.ensureCerts();\n\n    const description = this.buildAnswer();\n    return description.toJSON();\n  }\n\n  private buildAnswer() {\n    this.assertNotClosed();\n    if (\n      ![\"have-remote-offer\", \"have-local-pranswer\"].includes(\n        this.signalingState,\n      )\n    ) {\n      throw new Error(\"createAnswer failed\");\n    }\n    if (!this._remoteDescription) {\n      throw new Error(\"wrong state\");\n    }\n\n    const description = new SessionDescription();\n    addSDPHeader(\"answer\", description);\n\n    this._remoteDescription.media.forEach((remoteMedia) => {\n      let dtlsTransport!: RTCDtlsTransport;\n      let media: MediaDescription;\n\n      if ([\"audio\", \"video\"].includes(remoteMedia.kind)) {\n        const transceiver = this.getTransceiverByMid(remoteMedia.rtp.muxId!)!;\n        media = createMediaDescriptionForTransceiver(\n          transceiver,\n          this.cname,\n          andDirection(transceiver.direction, transceiver.offerDirection),\n        );\n        dtlsTransport = transceiver.dtlsTransport;\n      } else if (remoteMedia.kind === \"application\") {\n        if (!this.sctpTransport || !this.sctpTransport.mid) {\n          throw new Error(\"sctpTransport not found\");\n        }\n        media = createMediaDescriptionForSctp(this.sctpTransport);\n\n        dtlsTransport = this.sctpTransport.dtlsTransport;\n      } else {\n        throw new Error(\"invalid kind\");\n      }\n\n      // # determine DTLS role, or preserve the currently configured role\n      if (media.dtlsParams) {\n        if (dtlsTransport.role === \"auto\") {\n          media.dtlsParams.role = \"client\";\n        } else {\n          media.dtlsParams.role = dtlsTransport.role;\n        }\n      }\n\n      media.simulcastParameters = remoteMedia.simulcastParameters.map((v) => ({\n        ...v,\n        direction: reverseSimulcastDirection(v.direction),\n      }));\n\n      description.media.push(media);\n    });\n\n    if (this.config.bundlePolicy !== \"disable\") {\n      const bundle = new GroupDescription(\"BUNDLE\", []);\n      for (const media of description.media) {\n        if (media.direction !== \"inactive\") {\n          bundle.items.push(media.rtp.muxId!);\n        }\n      }\n      description.group.push(bundle);\n    }\n\n    return description;\n  }\n\n  async close() {\n    if (this.isClosed) return;\n\n    this.isClosed = true;\n    this.setSignalingState(\"closed\");\n    this.setConnectionState(\"closed\");\n\n    this.transceivers.forEach((transceiver) => {\n      transceiver.receiver.stop();\n      transceiver.sender.stop();\n    });\n\n    if (this.sctpTransport) {\n      await this.sctpTransport.stop();\n    }\n    for (const dtlsTransport of this.dtlsTransports) {\n      await dtlsTransport.stop();\n      await dtlsTransport.iceTransport.stop();\n    }\n\n    this.dispose();\n    log(\"peerConnection closed\");\n  }\n\n  private assertNotClosed() {\n    if (this.isClosed) {\n      throw new Error(\"RTCPeerConnection is closed\");\n    }\n  }\n\n  // https://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate\n  private updateIceGatheringState() {\n    const all = this.iceTransports;\n\n    function allMatch(...state: IceGathererState[]) {\n      return (\n        all.filter((check) => state.includes(check.gatheringState)).length ===\n        all.length\n      );\n    }\n\n    let newState: IceGathererState;\n\n    if (all.length && allMatch(\"complete\")) {\n      newState = \"complete\";\n    } else if (!all.length || allMatch(\"new\", \"complete\")) {\n      newState = \"new\";\n    } else if (all.map((check) => check.gatheringState).includes(\"gathering\")) {\n      newState = \"gathering\";\n    } else {\n      newState = \"new\";\n    }\n\n    if (this.iceGatheringState === newState) {\n      return;\n    }\n\n    log(\"iceGatheringStateChange\", newState);\n    this.iceGatheringState = newState;\n    this.iceGatheringStateChange.execute(newState);\n    this.emit(\"icegatheringstatechange\", newState);\n  }\n\n  // https://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate\n  private updateIceConnectionState() {\n    const all = this.iceTransports;\n    let newState: RTCIceConnectionState;\n\n    function allMatch(...state: RTCIceConnectionState[]) {\n      return (\n        all.filter((check) => state.includes(check.state)).length === all.length\n      );\n    }\n\n    if (this.connectionState === \"closed\") {\n      newState = \"closed\";\n    } else if (allMatch(\"failed\")) {\n      newState = \"failed\";\n    } else if (allMatch(\"disconnected\")) {\n      newState = \"disconnected\";\n    } else if (allMatch(\"new\", \"closed\")) {\n      newState = \"new\";\n    } else if (allMatch(\"new\", \"checking\")) {\n      newState = \"checking\";\n    } else if (allMatch(\"completed\", \"closed\")) {\n      newState = \"completed\";\n    } else if (allMatch(\"connected\", \"completed\", \"closed\")) {\n      newState = \"connected\";\n    } else {\n      // unreachable?\n      newState = \"new\";\n    }\n\n    if (this.iceConnectionState === newState) {\n      return;\n    }\n\n    log(\"iceConnectionStateChange\", newState);\n    this.iceConnectionState = newState;\n    this.iceConnectionStateChange.execute(newState);\n    this.emit(\"iceconnectionstatechange\", newState);\n  }\n\n  private setSignalingState(state: RTCSignalingState) {\n    log(\"signalingStateChange\", state);\n    this.signalingState = state;\n    this.signalingStateChange.execute(state);\n    if (this.onsignalingstatechange) this.onsignalingstatechange({});\n  }\n\n  private setConnectionState(state: ConnectionState) {\n    log(\"connectionStateChange\", state);\n    this.connectionState = state;\n    this.connectionStateChange.execute(state);\n    if (this.onconnectionstatechange) {\n      this.onconnectionstatechange();\n    }\n    this.emit(\"connectionstatechange\");\n  }\n\n  private dispose() {\n    this.onDataChannel.allUnsubscribe();\n    this.iceGatheringStateChange.allUnsubscribe();\n    this.iceConnectionStateChange.allUnsubscribe();\n    this.signalingStateChange.allUnsubscribe();\n    this.onTransceiverAdded.allUnsubscribe();\n    this.onRemoteTransceiverAdded.allUnsubscribe();\n    this.onIceCandidate.allUnsubscribe();\n  }\n}\n\nexport function createMediaDescriptionForTransceiver(\n  transceiver: RTCRtpTransceiver,\n  cname: string,\n  direction: Direction,\n) {\n  const media = new MediaDescription(\n    transceiver.kind,\n    9,\n    \"UDP/TLS/RTP/SAVPF\",\n    transceiver.codecs.map((c) => c.payloadType),\n  );\n  media.direction = direction;\n  media.msid = transceiver.msid;\n  media.rtp = {\n    codecs: transceiver.codecs,\n    headerExtensions: transceiver.headerExtensions,\n    muxId: transceiver.mid,\n  };\n  media.rtcpHost = \"0.0.0.0\";\n  media.rtcpPort = 9;\n  media.rtcpMux = true;\n  media.ssrc = [new SsrcDescription({ ssrc: transceiver.sender.ssrc, cname })];\n\n  if (transceiver.options.simulcast) {\n    media.simulcastParameters = transceiver.options.simulcast.map(\n      (o) => new RTCRtpSimulcastParameters(o),\n    );\n  }\n\n  if (media.rtp.codecs.find((c) => c.name.toLowerCase() === \"rtx\")) {\n    media.ssrc.push(\n      new SsrcDescription({ ssrc: transceiver.sender.rtxSsrc, cname }),\n    );\n    media.ssrcGroup = [\n      new GroupDescription(\"FID\", [\n        transceiver.sender.ssrc.toString(),\n        transceiver.sender.rtxSsrc.toString(),\n      ]),\n    ];\n  }\n\n  addTransportDescription(media, transceiver.dtlsTransport);\n  return media;\n}\n\nexport function createMediaDescriptionForSctp(sctp: RTCSctpTransport) {\n  const media = new MediaDescription(\n    \"application\",\n    DISCARD_PORT,\n    \"UDP/DTLS/SCTP\",\n    [\"webrtc-datachannel\"],\n  );\n  media.sctpPort = sctp.port;\n  media.rtp.muxId = sctp.mid;\n  media.sctpCapabilities = RTCSctpTransport.getCapabilities();\n\n  addTransportDescription(media, sctp.dtlsTransport);\n  return media;\n}\n\nexport function addTransportDescription(\n  media: MediaDescription,\n  dtlsTransport: RTCDtlsTransport,\n) {\n  const iceTransport = dtlsTransport.iceTransport;\n\n  media.iceCandidates = iceTransport.localCandidates;\n  media.iceCandidatesComplete = iceTransport.gatheringState === \"complete\";\n  media.iceParams = iceTransport.localParameters;\n  media.iceOptions = \"trickle\";\n\n  media.host = DISCARD_HOST;\n  media.port = DISCARD_PORT;\n\n  if (media.direction === \"inactive\") {\n    media.port = 0;\n    media.msid = undefined;\n  }\n\n  if (!media.dtlsParams) {\n    media.dtlsParams = dtlsTransport.localParameters;\n    if (!media.dtlsParams.fingerprints) {\n      media.dtlsParams.fingerprints =\n        dtlsTransport.localParameters.fingerprints;\n    }\n  }\n}\n\nexport function allocateMid(mids: Set<string>, type: \"dc\" | \"av\" | \"\") {\n  let mid = \"\";\n  for (let i = 0; ; ) {\n    // rfc9143.html#name-security-considerations\n    // SHOULD be 3 bytes or fewer to allow them to efficiently fit into the MID RTP header extension\n    mid = (i++).toString() + type;\n    if (!mids.has(mid)) break;\n  }\n  mids.add(mid);\n  return mid;\n}\n\nexport type BundlePolicy = \"max-compat\" | \"max-bundle\" | \"disable\";\n\nexport interface PeerConfig {\n  codecs: Partial<{\n    /**\n     * When specifying a codec with a fixed payloadType such as PCMU,\n     * it is necessary to set the correct PayloadType in RTCRtpCodecParameters in advance.\n     */\n    audio: RTCRtpCodecParameters[];\n    video: RTCRtpCodecParameters[];\n  }>;\n  headerExtensions: Partial<{\n    audio: RTCRtpHeaderExtensionParameters[];\n    video: RTCRtpHeaderExtensionParameters[];\n  }>;\n  iceTransportPolicy: \"all\" | \"relay\";\n  iceServers: RTCIceServer[];\n  /**Minimum port and Maximum port must not be the same value */\n  icePortRange: [number, number] | undefined;\n  iceInterfaceAddresses: InterfaceAddresses | undefined;\n  /** Add additional host (local) addresses to use for candidate gathering.\n   * Notably, you can include hosts that are normally excluded, such as loopback, tun interfaces, etc.\n   */\n  iceAdditionalHostAddresses: string[] | undefined;\n  iceUseIpv4: boolean;\n  iceUseIpv6: boolean;\n  forceTurnTCP: boolean;\n  /** If provided, is called on each STUN request.\n   * Return `true` if a STUN response should be sent, false if it should be skipped. */\n  iceFilterStunResponse:\n    | ((message: Message, addr: Address, protocol: Protocol) => boolean)\n    | undefined;\n  iceFilterCandidatePair: ((pair: CandidatePair) => boolean) | undefined;\n  dtls: Partial<{\n    keys: DtlsKeys;\n  }>;\n  icePasswordPrefix: string | undefined;\n  bundlePolicy: BundlePolicy;\n  debug: Partial<{\n    /**% */\n    inboundPacketLoss: number;\n    /**% */\n    outboundPacketLoss: number;\n    /**ms */\n    receiverReportDelay: number;\n    disableSendNack: boolean;\n    disableRecvRetransmit: boolean;\n  }>;\n  midSuffix: boolean;\n}\n\nexport const findCodecByMimeType = (\n  codecs: RTCRtpCodecParameters[],\n  target: RTCRtpCodecParameters,\n) =>\n  codecs.find(\n    (localCodec) =>\n      localCodec.mimeType.toLowerCase() === target.mimeType.toLowerCase(),\n  )\n    ? target\n    : undefined;\n\nexport type RTCIceServer = {\n  urls: string;\n  username?: string;\n  credential?: string;\n};\n\nexport const defaultPeerConfig: PeerConfig = {\n  codecs: {\n    audio: [\n      new RTCRtpCodecParameters({\n        mimeType: \"audio/opus\",\n        clockRate: 48000,\n        channels: 2,\n      }),\n      new RTCRtpCodecParameters({\n        mimeType: \"audio/PCMU\",\n        clockRate: 8000,\n        channels: 1,\n        payloadType: 0,\n      }),\n    ],\n    video: [\n      new RTCRtpCodecParameters({\n        mimeType: \"video/VP8\",\n        clockRate: 90000,\n        rtcpFeedback: [useNACK(), usePLI(), useREMB()],\n      }),\n    ],\n  },\n  headerExtensions: {\n    audio: [],\n    video: [],\n  },\n  iceTransportPolicy: \"all\",\n  iceServers: [{ urls: \"stun:stun.l.google.com:19302\" }],\n  icePortRange: undefined,\n  iceInterfaceAddresses: undefined,\n  iceAdditionalHostAddresses: undefined,\n  iceUseIpv4: true,\n  iceUseIpv6: true,\n  iceFilterStunResponse: undefined,\n  iceFilterCandidatePair: undefined,\n  icePasswordPrefix: undefined,\n  dtls: {},\n  bundlePolicy: \"max-compat\",\n  debug: {},\n  midSuffix: false,\n  forceTurnTCP: false,\n};\n\nexport interface RTCTrackEvent {\n  track: MediaStreamTrack;\n  streams: MediaStream[];\n  transceiver: RTCRtpTransceiver;\n  receiver: RTCRtpReceiver;\n}\n\nexport interface RTCDataChannelEvent {\n  channel: RTCDataChannel;\n}\n\nexport interface RTCPeerConnectionIceEvent {\n  candidate?: RTCIceCandidate;\n}\n\ntype Media = \"audio\" | \"video\";\n\nexport interface RTCSessionDescriptionInit {\n  sdp?: string;\n  type: RTCSdpType;\n}\nexport type RTCSdpType = \"answer\" | \"offer\" | \"pranswer\" | \"rollback\";\n"]}
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debug from \"debug\";\nimport cloneDeep from \"lodash/cloneDeep\";\nimport isEqual from \"lodash/isEqual\";\nimport * as uuid from \"uuid\";\nimport { Event } from \"./imports/common\";\n\nimport {\n  type Address,\n  type CandidatePair,\n  type InterfaceAddresses,\n  Recvonly,\n  Sendonly,\n  Sendrecv,\n  deepMerge,\n} from \".\";\nimport {\n  type DtlsKeys,\n  codecParametersFromString,\n  useNACK,\n  usePLI,\n  useREMB,\n} from \".\";\nimport type { Profile } from \"../../dtls/src/context/srtp\";\nimport type { Message } from \"../../ice/src/stun/message\";\nimport type { Protocol } from \"../../ice/src/types/model\";\nimport {\n  DISCARD_HOST,\n  DISCARD_PORT,\n  ReceiverDirection,\n  SRTP_PROFILE,\n  SenderDirections,\n} from \"./const\";\nimport { RTCDataChannel, RTCDataChannelParameters } from \"./dataChannel\";\nimport { EventTarget, enumerate } from \"./helper\";\nimport {\n  RTCRtpCodecParameters,\n  RTCRtpCodingParameters,\n  type RTCRtpHeaderExtensionParameters,\n  type RTCRtpParameters,\n  type RTCRtpReceiveParameters,\n  RTCRtpRtxParameters,\n  RTCRtpSimulcastParameters,\n} from \"./media/parameters\";\nimport { RtpRouter } from \"./media/router\";\nimport { RTCRtpReceiver } from \"./media/rtpReceiver\";\nimport { RTCRtpSender } from \"./media/rtpSender\";\nimport {\n  type Direction,\n  RTCRtpTransceiver,\n  type TransceiverOptions,\n} from \"./media/rtpTransceiver\";\nimport { MediaStream, type MediaStreamTrack } from \"./media/track\";\nimport {\n  GroupDescription,\n  MediaDescription,\n  SessionDescription,\n  SsrcDescription,\n  addSDPHeader,\n} from \"./sdp\";\nimport { RTCCertificate, RTCDtlsTransport } from \"./transport/dtls\";\nimport {\n  IceCandidate,\n  type IceGathererState,\n  type RTCIceCandidate,\n  type RTCIceConnectionState,\n  RTCIceGatherer,\n  RTCIceTransport,\n} from \"./transport/ice\";\nimport { RTCSctpTransport } from \"./transport/sctp\";\nimport type { ConnectionState, Kind, RTCSignalingState } from \"./types/domain\";\nimport type { Callback, CallbackWithValue } from \"./types/util\";\nimport {\n  andDirection,\n  parseIceServers,\n  reverseDirection,\n  reverseSimulcastDirection,\n} from \"./utils\";\n\nconst log = debug(\"werift:packages/webrtc/src/peerConnection.ts\");\n\nexport class RTCPeerConnection extends EventTarget {\n  readonly cname = uuid.v4();\n  sctpTransport?: RTCSctpTransport;\n  config: Required<PeerConfig> = cloneDeep<PeerConfig>(defaultPeerConfig);\n  connectionState: ConnectionState = \"new\";\n  iceConnectionState: RTCIceConnectionState = \"new\";\n  iceGatheringState: IceGathererState = \"new\";\n  signalingState: RTCSignalingState = \"stable\";\n  negotiationneeded = false;\n  private readonly transceivers: RTCRtpTransceiver[] = [];\n  private pushTransceiver(t: RTCRtpTransceiver) {\n    this.transceivers.push(t);\n  }\n  private replaceTransceiver(t: RTCRtpTransceiver, index: number) {\n    this.transceivers[index] = t;\n  }\n  needRestart = false;\n\n  readonly iceGatheringStateChange = new Event<[IceGathererState]>();\n  readonly iceConnectionStateChange = new Event<[RTCIceConnectionState]>();\n  readonly signalingStateChange = new Event<[RTCSignalingState]>();\n  readonly connectionStateChange = new Event<[ConnectionState]>();\n  readonly onDataChannel = new Event<[RTCDataChannel]>();\n  readonly onRemoteTransceiverAdded = new Event<[RTCRtpTransceiver]>();\n  readonly onTransceiverAdded = new Event<[RTCRtpTransceiver]>();\n  readonly onIceCandidate = new Event<[RTCIceCandidate | undefined]>();\n  readonly onNegotiationneeded = new Event<[]>();\n  readonly onTrack = new Event<[MediaStreamTrack]>();\n\n  ondatachannel?: CallbackWithValue<RTCDataChannelEvent>;\n  onicecandidate?: CallbackWithValue<RTCPeerConnectionIceEvent>;\n  onnegotiationneeded?: CallbackWithValue<any>;\n  onsignalingstatechange?: CallbackWithValue<any>;\n  ontrack?: CallbackWithValue<RTCTrackEvent>;\n  onconnectionstatechange?: Callback;\n\n  private readonly router = new RtpRouter();\n  private certificate?: RTCCertificate;\n  sctpRemotePort?: number;\n  private seenMid = new Set<string>();\n  private currentLocalDescription?: SessionDescription;\n  private currentRemoteDescription?: SessionDescription;\n  private pendingLocalDescription?: SessionDescription;\n  private pendingRemoteDescription?: SessionDescription;\n  private isClosed = false;\n  private shouldNegotiationneeded = false;\n\n  get dtlsTransports() {\n    const transports = this.transceivers.map((t) => t.dtlsTransport);\n    if (this.sctpTransport) {\n      transports.push(this.sctpTransport.dtlsTransport);\n    }\n    return transports.reduce((acc: RTCDtlsTransport[], cur) => {\n      if (!acc.map((d) => d.id).includes(cur.id)) {\n        acc.push(cur);\n      }\n      return acc;\n    }, []);\n  }\n  get iceTransports() {\n    return this.dtlsTransports.map((d) => d.iceTransport);\n  }\n\n  get extIdUriMap() {\n    return this.router.extIdUriMap;\n  }\n\n  get iceGeneration() {\n    return this.iceTransports[0].connection.generation;\n  }\n\n  constructor(config: Partial<PeerConfig> = {}) {\n    super();\n\n    this.setConfiguration(config);\n\n    this.iceConnectionStateChange.subscribe((state) => {\n      switch (state) {\n        case \"disconnected\":\n          this.setConnectionState(\"disconnected\");\n          break;\n        case \"closed\":\n          this.close();\n          break;\n      }\n    });\n  }\n\n  setConfiguration(config: Partial<PeerConfig>) {\n    deepMerge(this.config, config);\n\n    if (this.config.icePortRange) {\n      const [min, max] = this.config.icePortRange;\n      if (min === max) throw new Error(\"should not be same value\");\n      if (min >= max) throw new Error(\"The min must be less than max\");\n    }\n\n    for (const [i, codecParams] of enumerate([\n      ...(this.config.codecs.audio || []),\n      ...(this.config.codecs.video || []),\n    ])) {\n      if (codecParams.payloadType != undefined) {\n        continue;\n      }\n\n      codecParams.payloadType = 96 + i;\n      switch (codecParams.name.toLowerCase()) {\n        case \"rtx\":\n          {\n            codecParams.parameters = `apt=${codecParams.payloadType - 1}`;\n          }\n          break;\n        case \"red\":\n          {\n            if (codecParams.contentType === \"audio\") {\n              const redundant = codecParams.payloadType + 1;\n              codecParams.parameters = `${redundant}/${redundant}`;\n              codecParams.payloadType = 63;\n            }\n          }\n          break;\n      }\n    }\n\n    [\n      ...(this.config.headerExtensions.audio || []),\n      ...(this.config.headerExtensions.video || []),\n    ].forEach((v, i) => {\n      v.id = 1 + i;\n    });\n\n    if (this.config.dtls) {\n      const { keys } = this.config.dtls;\n\n      if (keys) {\n        this.certificate = new RTCCertificate(\n          keys.keyPem,\n          keys.certPem,\n          keys.signatureHash,\n        );\n      }\n    }\n  }\n\n  get localDescription() {\n    if (!this._localDescription) {\n      return undefined;\n    }\n    return this._localDescription.toJSON();\n  }\n\n  get remoteDescription() {\n    if (!this._remoteDescription) {\n      return undefined;\n    }\n    return this._remoteDescription.toJSON();\n  }\n\n  /**@private */\n  get _localDescription() {\n    return this.pendingLocalDescription || this.currentLocalDescription;\n  }\n\n  /**@private */\n  get _remoteDescription() {\n    return this.pendingRemoteDescription || this.currentRemoteDescription;\n  }\n\n  private getTransceiverByMid(mid: string) {\n    return this.transceivers.find((transceiver) => transceiver.mid === mid);\n  }\n\n  private getTransceiverByMLineIndex(index: number) {\n    return this.transceivers.find(\n      (transceiver) => transceiver.mLineIndex === index,\n    );\n  }\n\n  getConfiguration() {\n    return this.config;\n  }\n\n  async createOffer({ iceRestart }: { iceRestart?: boolean } = {}) {\n    if (iceRestart || this.needRestart) {\n      this.needRestart = false;\n      for (const t of this.iceTransports) {\n        t.restart();\n      }\n    }\n\n    await this.ensureCerts();\n\n    const description = this.buildOfferSdp();\n    return description.toJSON();\n  }\n\n  private assignTransceiverCodecs(transceiver: RTCRtpTransceiver) {\n    const codecs = (\n      this.config.codecs[transceiver.kind] as RTCRtpCodecParameters[]\n    ).filter((codecCandidate) => {\n      switch (codecCandidate.direction) {\n        case \"recvonly\": {\n          if (ReceiverDirection.includes(transceiver.direction)) return true;\n          return false;\n        }\n        case \"sendonly\": {\n          if (SenderDirections.includes(transceiver.direction)) return true;\n          return false;\n        }\n        case \"sendrecv\": {\n          if ([Sendrecv, Recvonly, Sendonly].includes(transceiver.direction))\n            return true;\n          return false;\n        }\n        case \"all\": {\n          return true;\n        }\n        default:\n          return false;\n      }\n    });\n    transceiver.codecs = codecs;\n  }\n\n  buildOfferSdp() {\n    this.transceivers.forEach((transceiver) => {\n      if (transceiver.codecs.length === 0) {\n        this.assignTransceiverCodecs(transceiver);\n      }\n      if (transceiver.headerExtensions.length === 0) {\n        transceiver.headerExtensions =\n          this.config.headerExtensions[transceiver.kind] ?? [];\n      }\n    });\n\n    const description = new SessionDescription();\n    addSDPHeader(\"offer\", description);\n\n    // # handle existing transceivers / sctp\n\n    const currentMedia = this._localDescription\n      ? this._localDescription.media\n      : [];\n\n    currentMedia.forEach((m, i) => {\n      const mid = m.rtp.muxId;\n      if (!mid) {\n        log(\"mid missing\", m);\n        return;\n      }\n      if (m.kind === \"application\") {\n        if (!this.sctpTransport) {\n          throw new Error(\"sctpTransport not found\");\n        }\n        this.sctpTransport.mLineIndex = i;\n        description.media.push(\n          createMediaDescriptionForSctp(this.sctpTransport),\n        );\n      } else {\n        const transceiver = this.getTransceiverByMid(mid);\n        if (!transceiver) {\n          if (m.direction === \"inactive\") {\n            description.media.push(m);\n            return;\n          }\n          throw new Error(\"transceiver not found\");\n        }\n        transceiver.mLineIndex = i;\n        description.media.push(\n          createMediaDescriptionForTransceiver(\n            transceiver,\n            this.cname,\n            transceiver.direction,\n          ),\n        );\n      }\n    });\n\n    // # handle new transceivers / sctp\n    this.transceivers\n      .filter((t) => !description.media.find((m) => m.rtp.muxId === t.mid))\n      .forEach((transceiver) => {\n        if (transceiver.mid == undefined) {\n          transceiver.mid = allocateMid(\n            this.seenMid,\n            this.config.midSuffix ? \"av\" : \"\",\n          );\n        }\n        const mediaDescription = createMediaDescriptionForTransceiver(\n          transceiver,\n          this.cname,\n          transceiver.direction,\n        );\n        if (transceiver.mLineIndex === undefined) {\n          transceiver.mLineIndex = description.media.length;\n          description.media.push(mediaDescription);\n        } else {\n          description.media[transceiver.mLineIndex] = mediaDescription;\n        }\n      });\n\n    if (\n      this.sctpTransport &&\n      !description.media.find((m) => m.kind === \"application\")\n    ) {\n      this.sctpTransport.mLineIndex = description.media.length;\n      if (this.sctpTransport.mid == undefined) {\n        this.sctpTransport.mid = allocateMid(\n          this.seenMid,\n          this.config.midSuffix ? \"dc\" : \"\",\n        );\n      }\n      description.media.push(createMediaDescriptionForSctp(this.sctpTransport));\n    }\n\n    if (this.config.bundlePolicy !== \"disable\") {\n      const mids = description.media\n        .map((m) => (m.direction !== \"inactive\" ? m.rtp.muxId : undefined))\n        .filter((v) => v) as string[];\n      if (mids.length) {\n        const bundle = new GroupDescription(\"BUNDLE\", mids);\n        description.group.push(bundle);\n      }\n    }\n\n    return description;\n  }\n\n  createDataChannel(\n    label: string,\n    options: Partial<{\n      maxPacketLifeTime?: number;\n      protocol: string;\n      maxRetransmits?: number;\n      ordered: boolean;\n      negotiated: boolean;\n      id?: number;\n    }> = {},\n  ): RTCDataChannel {\n    const base: typeof options = {\n      protocol: \"\",\n      ordered: true,\n      negotiated: false,\n    };\n    const settings: Required<typeof base> = { ...base, ...options } as any;\n\n    if (settings.maxPacketLifeTime && settings.maxRetransmits) {\n      throw new Error(\"can not select both\");\n    }\n\n    if (!this.sctpTransport) {\n      this.sctpTransport = this.createSctpTransport();\n      this.needNegotiation();\n    }\n\n    const parameters = new RTCDataChannelParameters({\n      id: settings.id,\n      label,\n      maxPacketLifeTime: settings.maxPacketLifeTime,\n      maxRetransmits: settings.maxRetransmits,\n      negotiated: settings.negotiated,\n      ordered: settings.ordered,\n      protocol: settings.protocol,\n    });\n\n    const channel = new RTCDataChannel(this.sctpTransport, parameters);\n    return channel;\n  }\n\n  removeTrack(sender: RTCRtpSender) {\n    if (this.isClosed) throw new Error(\"peer closed\");\n    if (!this.getSenders().find(({ ssrc }) => sender.ssrc === ssrc)) {\n      throw new Error(\"unExist\");\n    }\n\n    const transceiver = this.transceivers.find(\n      ({ sender: { ssrc } }) => sender.ssrc === ssrc,\n    );\n    if (!transceiver) throw new Error(\"unExist\");\n\n    sender.stop();\n\n    if (transceiver.currentDirection === \"recvonly\") {\n      this.needNegotiation();\n      return;\n    }\n\n    if (transceiver.stopping || transceiver.stopped) {\n      transceiver.setDirection(\"inactive\");\n    } else {\n      if (transceiver.direction === \"sendrecv\") {\n        transceiver.setDirection(\"recvonly\");\n      } else if (\n        transceiver.direction === \"sendonly\" ||\n        transceiver.direction === \"recvonly\"\n      ) {\n        transceiver.setDirection(\"inactive\");\n      }\n    }\n    this.needNegotiation();\n  }\n\n  private needNegotiation = async () => {\n    this.shouldNegotiationneeded = true;\n    if (this.negotiationneeded || this.signalingState !== \"stable\") {\n      return;\n    }\n    this.shouldNegotiationneeded = false;\n    setImmediate(() => {\n      this.negotiationneeded = true;\n      this.onNegotiationneeded.execute();\n      if (this.onnegotiationneeded) this.onnegotiationneeded({});\n    });\n  };\n\n  private createTransport(srtpProfiles: Profile[] = []) {\n    const [existing] = this.iceTransports;\n\n    // Gather ICE candidates for only one track. If the remote endpoint is not bundle-aware, negotiate only one media track.\n    // https://w3c.github.io/webrtc-pc/#rtcbundlepolicy-enum\n    if (this.config.bundlePolicy === \"max-bundle\") {\n      if (existing) {\n        return this.dtlsTransports[0];\n      }\n    }\n\n    const iceGatherer = new RTCIceGatherer({\n      ...parseIceServers(this.config.iceServers),\n      forceTurn: this.config.iceTransportPolicy === \"relay\",\n      portRange: this.config.icePortRange,\n      interfaceAddresses: this.config.iceInterfaceAddresses,\n      additionalHostAddresses: this.config.iceAdditionalHostAddresses,\n      filterStunResponse: this.config.iceFilterStunResponse,\n      filterCandidatePair: this.config.iceFilterCandidatePair,\n      localPasswordPrefix: this.config.icePasswordPrefix,\n      useIpv4: this.config.iceUseIpv4,\n      useIpv6: this.config.iceUseIpv6,\n      turnTransport: this.config.forceTurnTCP === true ? \"tcp\" : \"udp\",\n    });\n    if (existing) {\n      iceGatherer.connection.localUsername = existing.connection.localUsername;\n      iceGatherer.connection.localPassword = existing.connection.localPassword;\n    }\n    iceGatherer.onGatheringStateChange.subscribe(() => {\n      this.updateIceGatheringState();\n    });\n    this.updateIceGatheringState();\n    const iceTransport = new RTCIceTransport(iceGatherer);\n    iceTransport.onStateChange.subscribe(() => {\n      this.updateIceConnectionState();\n    });\n    iceTransport.onNegotiationNeeded.subscribe(() => {\n      this.needNegotiation();\n    });\n    iceTransport.onIceCandidate.subscribe((candidate) => {\n      if (!this.localDescription) {\n        log(\"localDescription not found when ice candidate was gathered\");\n        return;\n      }\n      if (!candidate) {\n        this.setLocal(this._localDescription!);\n        this.onIceCandidate.execute(undefined);\n        if (this.onicecandidate) {\n          this.onicecandidate({ candidate: undefined });\n        }\n        this.emit(\"icecandidate\", { candidate: undefined });\n        return;\n      }\n\n      if (this.config.bundlePolicy === \"max-bundle\" || this.remoteIsBundled) {\n        candidate.sdpMLineIndex = 0;\n        const media = this._localDescription?.media[0];\n        if (media) {\n          candidate.sdpMid = media.rtp.muxId;\n        }\n      } else {\n        const transceiver = this.transceivers.find(\n          (t) => t.dtlsTransport.iceTransport.id === iceTransport.id,\n        );\n        if (transceiver) {\n          candidate.sdpMLineIndex = transceiver.mLineIndex;\n          candidate.sdpMid = transceiver.mid;\n        }\n        if (\n          this.sctpTransport?.dtlsTransport.iceTransport.id === iceTransport.id\n        ) {\n          candidate.sdpMLineIndex = this.sctpTransport.mLineIndex;\n          candidate.sdpMid = this.sctpTransport.mid;\n        }\n      }\n\n      candidate.foundation = \"candidate:\" + candidate.foundation;\n\n      this.onIceCandidate.execute(candidate.toJSON());\n      if (this.onicecandidate) {\n        this.onicecandidate({ candidate: candidate.toJSON() });\n      }\n      this.emit(\"icecandidate\", { candidate });\n    });\n\n    const dtlsTransport = new RTCDtlsTransport(\n      this.config,\n      iceTransport,\n      this.router,\n      this.certificate,\n      srtpProfiles,\n    );\n\n    return dtlsTransport;\n  }\n\n  private createSctpTransport() {\n    const dtlsTransport = this.createTransport([\n      SRTP_PROFILE.SRTP_AEAD_AES_128_GCM, // prefer\n      SRTP_PROFILE.SRTP_AES128_CM_HMAC_SHA1_80,\n    ]);\n    const sctp = new RTCSctpTransport();\n    sctp.setDtlsTransport(dtlsTransport);\n    sctp.mid = undefined;\n\n    sctp.onDataChannel.subscribe((channel) => {\n      this.onDataChannel.execute(channel);\n\n      const event: RTCDataChannelEvent = { channel };\n      if (this.ondatachannel) this.ondatachannel(event);\n      this.emit(\"datachannel\", event);\n    });\n\n    this.sctpTransport = sctp;\n    this.updateIceConnectionState();\n\n    return sctp;\n  }\n\n  async setLocalDescription(sessionDescription: {\n    type: \"offer\" | \"answer\";\n    sdp: string;\n  }): Promise<SessionDescription> {\n    // # parse and validate description\n    const description = SessionDescription.parse(sessionDescription.sdp);\n    description.type = sessionDescription.type;\n    this.validateDescription(description, true);\n\n    // # update signaling state\n    if (description.type === \"offer\") {\n      this.setSignalingState(\"have-local-offer\");\n    } else if (description.type === \"answer\") {\n      this.setSignalingState(\"stable\");\n    }\n\n    // # assign MID\n    for (const [i, media] of enumerate(description.media)) {\n      const mid = media.rtp.muxId!;\n      this.seenMid.add(mid);\n      if ([\"audio\", \"video\"].includes(media.kind)) {\n        const transceiver = this.getTransceiverByMLineIndex(i);\n        if (transceiver) {\n          transceiver.mid = mid;\n        }\n      }\n      if (media.kind === \"application\" && this.sctpTransport) {\n        this.sctpTransport.mid = mid;\n      }\n    }\n\n    // setup ice,dtls role\n    for (const dtlsTransport of this.dtlsTransports) {\n      const iceTransport = dtlsTransport.iceTransport;\n\n      // # set ICE role\n      if (description.type === \"offer\") {\n        iceTransport.connection.iceControlling = true;\n      } else {\n        iceTransport.connection.iceControlling = false;\n      }\n      // One agent full, one lite:  The full agent MUST take the controlling role, and the lite agent MUST take the controlled role\n      // RFC 8445 S6.1.1\n      if (iceTransport.connection.remoteIsLite) {\n        iceTransport.connection.iceControlling = true;\n      }\n\n      // # set DTLS role for mediasoup\n      if (description.type === \"answer\") {\n        const role = description.media.find((media) => media.dtlsParams)\n          ?.dtlsParams?.role;\n        if (role) {\n          dtlsTransport.role = role;\n        }\n      }\n    }\n\n    // # configure direction\n    if ([\"answer\", \"pranswer\"].includes(description.type)) {\n      for (const t of this.transceivers) {\n        const direction = andDirection(t.direction, t.offerDirection);\n        t.setCurrentDirection(direction);\n      }\n    }\n\n    // for trickle ice\n    this.setLocal(description);\n\n    await this.gatherCandidates().catch((e) => {\n      log(\"gatherCandidates failed\", e);\n    });\n\n    // connect transports\n    if (description.type === \"answer\") {\n      this.connect().catch((err) => {\n        log(\"connect failed\", err);\n        this.setConnectionState(\"failed\");\n      });\n    }\n\n    this.setLocal(description);\n\n    if (this.shouldNegotiationneeded) {\n      this.needNegotiation();\n    }\n\n    return description;\n  }\n\n  private async gatherCandidates() {\n    // # gather candidates\n    const connected = this.iceTransports.find(\n      (transport) => transport.state === \"connected\",\n    );\n    if (this.remoteIsBundled && connected) {\n      // no need to gather ice candidates on an existing bundled connection\n    } else {\n      await Promise.allSettled(\n        this.iceTransports.map((iceTransport) => iceTransport.gather()),\n      );\n    }\n  }\n\n  private setLocal(description: SessionDescription) {\n    description.media\n      .filter((m) => [\"audio\", \"video\"].includes(m.kind))\n      .forEach((m, i) => {\n        addTransportDescription(m, this.transceivers[i].dtlsTransport);\n      });\n    const sctpMedia = description.media.find((m) => m.kind === \"application\");\n    if (this.sctpTransport && sctpMedia) {\n      addTransportDescription(sctpMedia, this.sctpTransport.dtlsTransport);\n    }\n\n    if (description.type === \"answer\") {\n      this.currentLocalDescription = description;\n      this.pendingLocalDescription = undefined;\n    } else {\n      this.pendingLocalDescription = description;\n    }\n  }\n\n  private getTransportByMid(mid: string) {\n    let iceTransport: RTCIceTransport | undefined;\n\n    const transceiver = this.transceivers.find((t) => t.mid === mid);\n    if (transceiver) {\n      iceTransport = transceiver.dtlsTransport.iceTransport;\n    } else if (!iceTransport && this.sctpTransport?.mid === mid) {\n      iceTransport = this.sctpTransport?.dtlsTransport.iceTransport;\n    }\n\n    return iceTransport;\n  }\n\n  private getTransportByMLineIndex(index: number) {\n    const sdp = this.buildOfferSdp();\n    const media = sdp.media[index];\n    if (!media) {\n      return;\n    }\n    const transport = this.getTransportByMid(media.rtp.muxId!);\n\n    return transport;\n  }\n\n  async addIceCandidate(candidateMessage: RTCIceCandidate) {\n    const candidate = IceCandidate.fromJSON(candidateMessage);\n    if (!candidate) {\n      return;\n    }\n\n    let iceTransport: RTCIceTransport | undefined;\n\n    if (typeof candidate.sdpMid === \"number\") {\n      iceTransport = this.getTransportByMid(candidate.sdpMid);\n    }\n\n    if (!iceTransport && typeof candidate.sdpMLineIndex === \"number\") {\n      iceTransport = this.getTransportByMLineIndex(candidate.sdpMLineIndex);\n    }\n\n    if (!iceTransport) {\n      iceTransport = this.iceTransports[0];\n    }\n\n    if (iceTransport) {\n      await iceTransport.addRemoteCandidate(candidate);\n    } else {\n      log(\"iceTransport not found\", candidate);\n    }\n  }\n\n  private async connect() {\n    log(\"start connect\");\n\n    await Promise.allSettled(\n      this.dtlsTransports.map(async (dtlsTransport) => {\n        const { iceTransport } = dtlsTransport;\n        if (iceTransport.state === \"connected\") {\n          return;\n        }\n\n        this.setConnectionState(\"connecting\");\n\n        await iceTransport.start().catch((err) => {\n          log(\"iceTransport.start failed\", err);\n          throw err;\n        });\n\n        if (dtlsTransport.state === \"connected\") {\n          return;\n        }\n        await dtlsTransport.start().catch((err) => {\n          log(\"dtlsTransport.start failed\", err);\n          throw err;\n        });\n        if (\n          this.sctpTransport &&\n          this.sctpRemotePort &&\n          this.sctpTransport.dtlsTransport.id === dtlsTransport.id\n        ) {\n          await this.sctpTransport.start(this.sctpRemotePort);\n          await this.sctpTransport.sctp.stateChanged.connected.asPromise();\n          log(\"sctp connected\");\n        }\n      }),\n    );\n\n    this.setConnectionState(\"connected\");\n  }\n\n  private getLocalRtpParams(transceiver: RTCRtpTransceiver): RTCRtpParameters {\n    if (transceiver.mid == undefined) throw new Error(\"mid not assigned\");\n\n    const rtp: RTCRtpParameters = {\n      codecs: transceiver.codecs,\n      muxId: transceiver.mid,\n      headerExtensions: transceiver.headerExtensions,\n      rtcp: { cname: this.cname, ssrc: transceiver.sender.ssrc, mux: true },\n    };\n    return rtp;\n  }\n\n  private getRemoteRtpParams(\n    media: MediaDescription,\n    transceiver: RTCRtpTransceiver,\n  ): RTCRtpReceiveParameters {\n    const receiveParameters: RTCRtpReceiveParameters = {\n      muxId: media.rtp.muxId,\n      rtcp: media.rtp.rtcp,\n      codecs: transceiver.codecs,\n      headerExtensions: transceiver.headerExtensions,\n      encodings: Object.values(\n        transceiver.codecs.reduce(\n          (acc: { [pt: number]: RTCRtpCodingParameters }, codec) => {\n            if (codec.name.toLowerCase() === \"rtx\") {\n              const params = codecParametersFromString(codec.parameters ?? \"\");\n              const apt = acc[params[\"apt\"]];\n              if (apt && media.ssrc.length === 2) {\n                apt.rtx = new RTCRtpRtxParameters({ ssrc: media.ssrc[1].ssrc });\n              }\n              return acc;\n            }\n            acc[codec.payloadType] = new RTCRtpCodingParameters({\n              ssrc: media.ssrc[0]?.ssrc,\n              payloadType: codec.payloadType,\n            });\n            return acc;\n          },\n          {},\n        ),\n      ),\n    };\n\n    return receiveParameters;\n  }\n\n  private get remoteIsBundled() {\n    const remoteSdp = this._remoteDescription;\n    if (!remoteSdp) {\n      return undefined;\n    }\n    const bundle = remoteSdp.group.find(\n      (g) => g.semantic === \"BUNDLE\" && this.config.bundlePolicy !== \"disable\",\n    );\n    return bundle;\n  }\n\n  restartIce() {\n    this.needRestart = true;\n    this.needNegotiation();\n  }\n\n  async setRemoteDescription(sessionDescription: RTCSessionDescriptionInit) {\n    if (\n      !sessionDescription.sdp ||\n      !sessionDescription.type ||\n      sessionDescription.type === \"rollback\" ||\n      sessionDescription.type === \"pranswer\"\n    ) {\n      throw new Error(\"invalid sessionDescription\");\n    }\n\n    // # parse and validate description\n    const remoteSdp = SessionDescription.parse(sessionDescription.sdp);\n    remoteSdp.type = sessionDescription.type;\n    this.validateDescription(remoteSdp, false);\n\n    if (remoteSdp.type === \"answer\") {\n      this.currentRemoteDescription = remoteSdp;\n      this.pendingRemoteDescription = undefined;\n    } else {\n      this.pendingRemoteDescription = remoteSdp;\n    }\n\n    let bundleTransport: RTCDtlsTransport | undefined;\n\n    // # apply description\n\n    const matchTransceiverWithMedia = (\n      transceiver: RTCRtpTransceiver,\n      media: MediaDescription,\n    ) =>\n      transceiver.kind === media.kind &&\n      [undefined, media.rtp.muxId].includes(transceiver.mid);\n\n    let transports = remoteSdp.media.map((remoteMedia, i) => {\n      let dtlsTransport: RTCDtlsTransport;\n\n      if ([\"audio\", \"video\"].includes(remoteMedia.kind)) {\n        let transceiver = this.transceivers.find((t) =>\n          matchTransceiverWithMedia(t, remoteMedia),\n        );\n        if (!transceiver) {\n          // create remote transceiver\n          transceiver = this._addTransceiver(remoteMedia.kind, {\n            direction: \"recvonly\",\n          });\n          transceiver.mid = remoteMedia.rtp.muxId;\n          this.onRemoteTransceiverAdded.execute(transceiver);\n        } else {\n          if (transceiver.direction === \"inactive\" && transceiver.stopping) {\n            transceiver.stopped = true;\n\n            if (sessionDescription.type === \"answer\") {\n              transceiver.setCurrentDirection(\"inactive\");\n            }\n            return;\n          }\n        }\n\n        if (this.remoteIsBundled) {\n          if (!bundleTransport) {\n            bundleTransport = transceiver.dtlsTransport;\n          } else {\n            transceiver.setDtlsTransport(bundleTransport);\n          }\n        }\n\n        dtlsTransport = transceiver.dtlsTransport;\n\n        this.setRemoteRTP(transceiver, remoteMedia, remoteSdp.type, i);\n      } else if (remoteMedia.kind === \"application\") {\n        if (!this.sctpTransport) {\n          this.sctpTransport = this.createSctpTransport();\n          this.sctpTransport.mid = remoteMedia.rtp.muxId;\n        }\n\n        if (this.remoteIsBundled) {\n          if (!bundleTransport) {\n            bundleTransport = this.sctpTransport.dtlsTransport;\n          } else {\n            this.sctpTransport.setDtlsTransport(bundleTransport);\n          }\n        }\n\n        dtlsTransport = this.sctpTransport.dtlsTransport;\n\n        this.setRemoteSCTP(remoteMedia, this.sctpTransport, i);\n      } else {\n        throw new Error(\"invalid media kind\");\n      }\n\n      const iceTransport = dtlsTransport.iceTransport;\n\n      if (remoteMedia.iceParams) {\n        const renomination = !!remoteSdp.media.find(\n          (m) => m.direction === \"inactive\",\n        );\n        iceTransport.setRemoteParams(remoteMedia.iceParams, renomination);\n\n        // One agent full, one lite:  The full agent MUST take the controlling role, and the lite agent MUST take the controlled role\n        // RFC 8445 S6.1.1\n        if (remoteMedia.iceParams?.iceLite) {\n          iceTransport.connection.iceControlling = true;\n        }\n      }\n      if (remoteMedia.dtlsParams) {\n        dtlsTransport.setRemoteParams(remoteMedia.dtlsParams);\n      }\n\n      // # add ICE candidates\n      remoteMedia.iceCandidates.forEach(iceTransport.addRemoteCandidate);\n\n      if (remoteMedia.iceCandidatesComplete) {\n        iceTransport.addRemoteCandidate(undefined);\n      }\n\n      // # set DTLS role\n      if (remoteSdp.type === \"answer\" && remoteMedia.dtlsParams?.role) {\n        dtlsTransport.role =\n          remoteMedia.dtlsParams.role === \"client\" ? \"server\" : \"client\";\n      }\n      return iceTransport;\n    }) as RTCIceTransport[];\n\n    // filter out inactive transports\n    transports = transports.filter((iceTransport) => !!iceTransport);\n\n    const removedTransceivers = this.transceivers.filter(\n      (t) =>\n        remoteSdp.media.find((m) => matchTransceiverWithMedia(t, m)) ==\n        undefined,\n    );\n\n    if (sessionDescription.type === \"answer\") {\n      for (const transceiver of removedTransceivers) {\n        // todo: handle answer side transceiver removal work.\n        // event should trigger to notify media source to stop.\n        transceiver.stop();\n        transceiver.stopped = true;\n      }\n    }\n\n    if (remoteSdp.type === \"offer\") {\n      this.setSignalingState(\"have-remote-offer\");\n    } else if (remoteSdp.type === \"answer\") {\n      this.setSignalingState(\"stable\");\n    }\n\n    // connect transports\n    if (remoteSdp.type === \"answer\") {\n      log(\"caller start connect\");\n      this.connect().catch((err) => {\n        log(\"connect failed\", err);\n        this.setConnectionState(\"failed\");\n      });\n    }\n\n    this.negotiationneeded = false;\n    if (this.shouldNegotiationneeded) {\n      this.needNegotiation();\n    }\n  }\n\n  private setRemoteRTP(\n    transceiver: RTCRtpTransceiver,\n    remoteMedia: MediaDescription,\n    type: \"offer\" | \"answer\",\n    mLineIndex: number,\n  ) {\n    if (!transceiver.mid) {\n      transceiver.mid = remoteMedia.rtp.muxId;\n    }\n    transceiver.mLineIndex = mLineIndex;\n\n    // # negotiate codecs\n    transceiver.codecs = remoteMedia.rtp.codecs.filter((remoteCodec) => {\n      const localCodecs = this.config.codecs[remoteMedia.kind] || [];\n\n      const existCodec = findCodecByMimeType(localCodecs, remoteCodec);\n      if (!existCodec) {\n        return false;\n      }\n\n      if (existCodec?.name.toLowerCase() === \"rtx\") {\n        const params = codecParametersFromString(existCodec.parameters ?? \"\");\n        const pt = params[\"apt\"];\n        const origin = remoteMedia.rtp.codecs.find((c) => c.payloadType === pt);\n        if (!origin) {\n          return false;\n        }\n        return !!findCodecByMimeType(localCodecs, origin);\n      }\n\n      return true;\n    });\n\n    log(\"negotiated codecs\", transceiver.codecs);\n    if (transceiver.codecs.length === 0) {\n      throw new Error(\"negotiate codecs failed.\");\n    }\n    transceiver.headerExtensions = remoteMedia.rtp.headerExtensions.filter(\n      (extension) =>\n        (this.config.headerExtensions[remoteMedia.kind as Media] || []).find(\n          (v) => v.uri === extension.uri,\n        ),\n    );\n\n    // # configure direction\n    const mediaDirection = remoteMedia.direction ?? \"inactive\";\n    const direction = reverseDirection(mediaDirection);\n    if ([\"answer\", \"pranswer\"].includes(type)) {\n      transceiver.setCurrentDirection(direction);\n    } else {\n      transceiver.offerDirection = direction;\n    }\n    const localParams = this.getLocalRtpParams(transceiver);\n    transceiver.sender.prepareSend(localParams);\n\n    if ([\"recvonly\", \"sendrecv\"].includes(transceiver.direction)) {\n      const remotePrams = this.getRemoteRtpParams(remoteMedia, transceiver);\n\n      // register simulcast receiver\n      remoteMedia.simulcastParameters.forEach((param) => {\n        this.router.registerRtpReceiverByRid(transceiver, param, remotePrams);\n      });\n\n      transceiver.receiver.prepareReceive(remotePrams);\n      // register ssrc receiver\n      this.router.registerRtpReceiverBySsrc(transceiver, remotePrams);\n    }\n    if ([\"sendonly\", \"sendrecv\"].includes(mediaDirection)) {\n      // assign msid\n      if (remoteMedia.msid != undefined) {\n        const [streamId, trackId] = remoteMedia.msid.split(\" \");\n        transceiver.receiver.remoteStreamId = streamId;\n        transceiver.receiver.remoteTrackId = trackId;\n\n        this.fireOnTrack(\n          transceiver.receiver.track,\n          transceiver,\n          new MediaStream({\n            id: streamId,\n            tracks: [transceiver.receiver.track],\n          }),\n        );\n      }\n    }\n\n    if (remoteMedia.ssrc[0]?.ssrc) {\n      transceiver.receiver.setupTWCC(remoteMedia.ssrc[0].ssrc);\n    }\n  }\n\n  private setRemoteSCTP(\n    remoteMedia: MediaDescription,\n    sctpTransport: RTCSctpTransport,\n    mLineIndex: number,\n  ) {\n    // # configure sctp\n    this.sctpRemotePort = remoteMedia.sctpPort;\n    if (!this.sctpRemotePort) {\n      throw new Error(\"sctpRemotePort not exist\");\n    }\n\n    sctpTransport.setRemotePort(this.sctpRemotePort);\n    sctpTransport.mLineIndex = mLineIndex;\n    if (!sctpTransport.mid) {\n      sctpTransport.mid = remoteMedia.rtp.muxId;\n    }\n  }\n\n  private validateDescription(\n    description: SessionDescription,\n    isLocal: boolean,\n  ) {\n    if (isLocal) {\n      if (description.type === \"offer\") {\n        if (![\"stable\", \"have-local-offer\"].includes(this.signalingState))\n          throw new Error(\"Cannot handle offer in signaling state\");\n      } else if (description.type === \"answer\") {\n        if (\n          ![\"have-remote-offer\", \"have-local-pranswer\"].includes(\n            this.signalingState,\n          )\n        ) {\n          throw new Error(\"Cannot handle answer in signaling state\");\n        }\n      }\n    } else {\n      if (description.type === \"offer\") {\n        if (![\"stable\", \"have-remote-offer\"].includes(this.signalingState)) {\n          throw new Error(\"Cannot handle offer in signaling state\");\n        }\n      } else if (description.type === \"answer\") {\n        if (\n          ![\"have-local-offer\", \"have-remote-pranswer\"].includes(\n            this.signalingState,\n          )\n        ) {\n          throw new Error(\"Cannot handle answer in signaling state\");\n        }\n      }\n    }\n\n    description.media.forEach((media) => {\n      if (media.direction === \"inactive\") return;\n      if (\n        !media.iceParams ||\n        !media.iceParams.usernameFragment ||\n        !media.iceParams.password\n      )\n        throw new Error(\"ICE username fragment or password is missing\");\n    });\n\n    if ([\"answer\", \"pranswer\"].includes(description.type || \"\")) {\n      const offer = isLocal ? this._remoteDescription : this._localDescription;\n      if (!offer) throw new Error();\n\n      const answerMedia = description.media.map((v, i) => [v.kind, i]);\n      const offerMedia = offer.media.map((v, i) => [v.kind, i]);\n      if (!isEqual(offerMedia, answerMedia)) {\n        throw new Error(\"Media sections in answer do not match offer\");\n      }\n    }\n  }\n\n  private fireOnTrack(\n    track: MediaStreamTrack,\n    transceiver: RTCRtpTransceiver,\n    stream: MediaStream,\n  ) {\n    const event: RTCTrackEvent = {\n      track,\n      streams: [stream],\n      transceiver,\n      receiver: transceiver.receiver,\n    };\n    this.onTrack.execute(track);\n    this.emit(\"track\", event);\n    if (this.ontrack) this.ontrack(event);\n  }\n\n  addTransceiver(\n    trackOrKind: Kind | MediaStreamTrack,\n    options: Partial<TransceiverOptions> = {},\n  ) {\n    return this._addTransceiver(trackOrKind, options);\n  }\n\n  private _addTransceiver(\n    trackOrKind: Kind | MediaStreamTrack,\n    options: Partial<TransceiverOptions> = {},\n  ) {\n    const kind =\n      typeof trackOrKind === \"string\" ? trackOrKind : trackOrKind.kind;\n\n    const direction = options.direction || \"sendrecv\";\n\n    const dtlsTransport = this.createTransport([\n      SRTP_PROFILE.SRTP_AEAD_AES_128_GCM, // prefer\n      SRTP_PROFILE.SRTP_AES128_CM_HMAC_SHA1_80,\n    ]);\n\n    const sender = new RTCRtpSender(trackOrKind);\n    const receiver = new RTCRtpReceiver(this.config, kind, sender.ssrc);\n    const newTransceiver = new RTCRtpTransceiver(\n      kind,\n      dtlsTransport,\n      receiver,\n      sender,\n      direction,\n    );\n    newTransceiver.options = options;\n    this.router.registerRtpSender(newTransceiver.sender);\n\n    // reuse inactive\n    const inactiveTransceiverIndex = this.transceivers.findIndex(\n      (t) => t.currentDirection === \"inactive\",\n    );\n    const inactiveTransceiver = this.transceivers.find(\n      (t) => t.currentDirection === \"inactive\",\n    );\n    if (inactiveTransceiverIndex > -1 && inactiveTransceiver) {\n      this.replaceTransceiver(newTransceiver, inactiveTransceiverIndex);\n      newTransceiver.mLineIndex = inactiveTransceiver.mLineIndex;\n      inactiveTransceiver.setCurrentDirection(undefined);\n    } else {\n      this.pushTransceiver(newTransceiver);\n    }\n    this.onTransceiverAdded.execute(newTransceiver);\n\n    this.updateIceConnectionState();\n    this.needNegotiation();\n\n    return newTransceiver;\n  }\n\n  getTransceivers() {\n    return this.transceivers;\n  }\n\n  getSenders(): RTCRtpSender[] {\n    return this.getTransceivers().map((t) => t.sender);\n  }\n\n  getReceivers() {\n    return this.getTransceivers().map((t) => t.receiver);\n  }\n\n  // todo fix\n  addTrack(\n    track: MediaStreamTrack,\n    /**todo impl */\n    ms?: MediaStream,\n  ) {\n    if (this.isClosed) {\n      throw new Error(\"is closed\");\n    }\n    if (this.getSenders().find((sender) => sender.track?.uuid === track.uuid)) {\n      throw new Error(\"track exist\");\n    }\n\n    const emptyTrackSender = this.transceivers.find(\n      (t) =>\n        t.sender.track == undefined &&\n        t.kind === track.kind &&\n        SenderDirections.includes(t.direction) === true,\n    );\n    if (emptyTrackSender) {\n      const sender = emptyTrackSender.sender;\n      sender.registerTrack(track);\n      this.needNegotiation();\n      return sender;\n    }\n\n    const notSendTransceiver = this.transceivers.find(\n      (t) =>\n        t.sender.track == undefined &&\n        t.kind === track.kind &&\n        SenderDirections.includes(t.direction) === false &&\n        !t.usedForSender,\n    );\n    if (notSendTransceiver) {\n      const sender = notSendTransceiver.sender;\n      sender.registerTrack(track);\n      switch (notSendTransceiver.direction) {\n        case \"recvonly\":\n          notSendTransceiver.setDirection(\"sendrecv\");\n          break;\n        case \"inactive\":\n          notSendTransceiver.setDirection(\"sendonly\");\n          break;\n      }\n      this.needNegotiation();\n      return sender;\n    } else {\n      const transceiver = this._addTransceiver(track, {\n        direction: \"sendrecv\",\n      });\n      this.needNegotiation();\n      return transceiver.sender;\n    }\n  }\n\n  private async ensureCerts() {\n    if (!this.certificate) {\n      this.certificate = await RTCDtlsTransport.SetupCertificate();\n    }\n\n    for (const dtlsTransport of this.dtlsTransports) {\n      dtlsTransport.localCertificate = this.certificate;\n    }\n  }\n\n  async createAnswer() {\n    await this.ensureCerts();\n\n    const description = this.buildAnswer();\n    return description.toJSON();\n  }\n\n  private buildAnswer() {\n    this.assertNotClosed();\n    if (\n      ![\"have-remote-offer\", \"have-local-pranswer\"].includes(\n        this.signalingState,\n      )\n    ) {\n      throw new Error(\"createAnswer failed\");\n    }\n    if (!this._remoteDescription) {\n      throw new Error(\"wrong state\");\n    }\n\n    const description = new SessionDescription();\n    addSDPHeader(\"answer\", description);\n\n    this._remoteDescription.media.forEach((remoteMedia) => {\n      let dtlsTransport!: RTCDtlsTransport;\n      let media: MediaDescription;\n\n      if ([\"audio\", \"video\"].includes(remoteMedia.kind)) {\n        const transceiver = this.getTransceiverByMid(remoteMedia.rtp.muxId!)!;\n        media = createMediaDescriptionForTransceiver(\n          transceiver,\n          this.cname,\n          andDirection(transceiver.direction, transceiver.offerDirection),\n        );\n        dtlsTransport = transceiver.dtlsTransport;\n      } else if (remoteMedia.kind === \"application\") {\n        if (!this.sctpTransport || !this.sctpTransport.mid) {\n          throw new Error(\"sctpTransport not found\");\n        }\n        media = createMediaDescriptionForSctp(this.sctpTransport);\n\n        dtlsTransport = this.sctpTransport.dtlsTransport;\n      } else {\n        throw new Error(\"invalid kind\");\n      }\n\n      // # determine DTLS role, or preserve the currently configured role\n      if (media.dtlsParams) {\n        if (dtlsTransport.role === \"auto\") {\n          media.dtlsParams.role = \"client\";\n        } else {\n          media.dtlsParams.role = dtlsTransport.role;\n        }\n      }\n\n      media.simulcastParameters = remoteMedia.simulcastParameters.map((v) => ({\n        ...v,\n        direction: reverseSimulcastDirection(v.direction),\n      }));\n\n      description.media.push(media);\n    });\n\n    if (this.config.bundlePolicy !== \"disable\") {\n      const bundle = new GroupDescription(\"BUNDLE\", []);\n      for (const media of description.media) {\n        if (media.direction !== \"inactive\") {\n          bundle.items.push(media.rtp.muxId!);\n        }\n      }\n      description.group.push(bundle);\n    }\n\n    return description;\n  }\n\n  async close() {\n    if (this.isClosed) return;\n\n    this.isClosed = true;\n    this.setSignalingState(\"closed\");\n    this.setConnectionState(\"closed\");\n\n    this.transceivers.forEach((transceiver) => {\n      transceiver.receiver.stop();\n      transceiver.sender.stop();\n    });\n\n    if (this.sctpTransport) {\n      await this.sctpTransport.stop();\n    }\n    for (const dtlsTransport of this.dtlsTransports) {\n      await dtlsTransport.stop();\n      await dtlsTransport.iceTransport.stop();\n    }\n\n    this.dispose();\n    log(\"peerConnection closed\");\n  }\n\n  private assertNotClosed() {\n    if (this.isClosed) {\n      throw new Error(\"RTCPeerConnection is closed\");\n    }\n  }\n\n  // https://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate\n  private updateIceGatheringState() {\n    const all = this.iceTransports;\n\n    function allMatch(...state: IceGathererState[]) {\n      return (\n        all.filter((check) => state.includes(check.gatheringState)).length ===\n        all.length\n      );\n    }\n\n    let newState: IceGathererState;\n\n    if (all.length && allMatch(\"complete\")) {\n      newState = \"complete\";\n    } else if (!all.length || allMatch(\"new\", \"complete\")) {\n      newState = \"new\";\n    } else if (all.map((check) => check.gatheringState).includes(\"gathering\")) {\n      newState = \"gathering\";\n    } else {\n      newState = \"new\";\n    }\n\n    if (this.iceGatheringState === newState) {\n      return;\n    }\n\n    log(\"iceGatheringStateChange\", newState);\n    this.iceGatheringState = newState;\n    this.iceGatheringStateChange.execute(newState);\n    this.emit(\"icegatheringstatechange\", newState);\n  }\n\n  // https://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate\n  private updateIceConnectionState() {\n    const all = this.iceTransports;\n    let newState: RTCIceConnectionState;\n\n    function allMatch(...state: RTCIceConnectionState[]) {\n      return (\n        all.filter((check) => state.includes(check.state)).length === all.length\n      );\n    }\n\n    if (this.connectionState === \"closed\") {\n      newState = \"closed\";\n    } else if (allMatch(\"failed\")) {\n      newState = \"failed\";\n    } else if (allMatch(\"disconnected\")) {\n      newState = \"disconnected\";\n    } else if (allMatch(\"new\", \"closed\")) {\n      newState = \"new\";\n    } else if (allMatch(\"new\", \"checking\")) {\n      newState = \"checking\";\n    } else if (allMatch(\"completed\", \"closed\")) {\n      newState = \"completed\";\n    } else if (allMatch(\"connected\", \"completed\", \"closed\")) {\n      newState = \"connected\";\n    } else {\n      // unreachable?\n      newState = \"new\";\n    }\n\n    if (this.iceConnectionState === newState) {\n      return;\n    }\n\n    log(\"iceConnectionStateChange\", newState);\n    this.iceConnectionState = newState;\n    this.iceConnectionStateChange.execute(newState);\n    this.emit(\"iceconnectionstatechange\", newState);\n  }\n\n  private setSignalingState(state: RTCSignalingState) {\n    log(\"signalingStateChange\", state);\n    this.signalingState = state;\n    this.signalingStateChange.execute(state);\n    if (this.onsignalingstatechange) this.onsignalingstatechange({});\n  }\n\n  private setConnectionState(state: ConnectionState) {\n    log(\"connectionStateChange\", state);\n    this.connectionState = state;\n    this.connectionStateChange.execute(state);\n    if (this.onconnectionstatechange) {\n      this.onconnectionstatechange();\n    }\n    this.emit(\"connectionstatechange\");\n  }\n\n  private dispose() {\n    this.onDataChannel.allUnsubscribe();\n    this.iceGatheringStateChange.allUnsubscribe();\n    this.iceConnectionStateChange.allUnsubscribe();\n    this.signalingStateChange.allUnsubscribe();\n    this.onTransceiverAdded.allUnsubscribe();\n    this.onRemoteTransceiverAdded.allUnsubscribe();\n    this.onIceCandidate.allUnsubscribe();\n  }\n}\n\nexport function createMediaDescriptionForTransceiver(\n  transceiver: RTCRtpTransceiver,\n  cname: string,\n  direction: Direction,\n) {\n  const media = new MediaDescription(\n    transceiver.kind,\n    9,\n    \"UDP/TLS/RTP/SAVPF\",\n    transceiver.codecs.map((c) => c.payloadType),\n  );\n  media.direction = direction;\n  media.msid = transceiver.msid;\n  media.rtp = {\n    codecs: transceiver.codecs,\n    headerExtensions: transceiver.headerExtensions,\n    muxId: transceiver.mid,\n  };\n  media.rtcpHost = \"0.0.0.0\";\n  media.rtcpPort = 9;\n  media.rtcpMux = true;\n  media.ssrc = [new SsrcDescription({ ssrc: transceiver.sender.ssrc, cname })];\n\n  if (transceiver.options.simulcast) {\n    media.simulcastParameters = transceiver.options.simulcast.map(\n      (o) => new RTCRtpSimulcastParameters(o),\n    );\n  }\n\n  if (media.rtp.codecs.find((c) => c.name.toLowerCase() === \"rtx\")) {\n    media.ssrc.push(\n      new SsrcDescription({ ssrc: transceiver.sender.rtxSsrc, cname }),\n    );\n    media.ssrcGroup = [\n      new GroupDescription(\"FID\", [\n        transceiver.sender.ssrc.toString(),\n        transceiver.sender.rtxSsrc.toString(),\n      ]),\n    ];\n  }\n\n  addTransportDescription(media, transceiver.dtlsTransport);\n  return media;\n}\n\nexport function createMediaDescriptionForSctp(sctp: RTCSctpTransport) {\n  const media = new MediaDescription(\n    \"application\",\n    DISCARD_PORT,\n    \"UDP/DTLS/SCTP\",\n    [\"webrtc-datachannel\"],\n  );\n  media.sctpPort = sctp.port;\n  media.rtp.muxId = sctp.mid;\n  media.sctpCapabilities = RTCSctpTransport.getCapabilities();\n\n  addTransportDescription(media, sctp.dtlsTransport);\n  return media;\n}\n\nexport function addTransportDescription(\n  media: MediaDescription,\n  dtlsTransport: RTCDtlsTransport,\n) {\n  const iceTransport = dtlsTransport.iceTransport;\n\n  media.iceCandidates = iceTransport.localCandidates;\n  media.iceCandidatesComplete = iceTransport.gatheringState === \"complete\";\n  media.iceParams = iceTransport.localParameters;\n  media.iceOptions = \"trickle\";\n\n  media.host = DISCARD_HOST;\n  media.port = DISCARD_PORT;\n\n  if (media.direction === \"inactive\") {\n    media.port = 0;\n    media.msid = undefined;\n  }\n\n  if (!media.dtlsParams) {\n    media.dtlsParams = dtlsTransport.localParameters;\n    if (!media.dtlsParams.fingerprints) {\n      media.dtlsParams.fingerprints =\n        dtlsTransport.localParameters.fingerprints;\n    }\n  }\n}\n\nexport function allocateMid(mids: Set<string>, type: \"dc\" | \"av\" | \"\") {\n  let mid = \"\";\n  for (let i = 0; ; ) {\n    // rfc9143.html#name-security-considerations\n    // SHOULD be 3 bytes or fewer to allow them to efficiently fit into the MID RTP header extension\n    mid = (i++).toString() + type;\n    if (!mids.has(mid)) break;\n  }\n  mids.add(mid);\n  return mid;\n}\n\nexport type BundlePolicy = \"max-compat\" | \"max-bundle\" | \"disable\";\n\nexport interface PeerConfig {\n  codecs: Partial<{\n    /**\n     * When specifying a codec with a fixed payloadType such as PCMU,\n     * it is necessary to set the correct PayloadType in RTCRtpCodecParameters in advance.\n     */\n    audio: RTCRtpCodecParameters[];\n    video: RTCRtpCodecParameters[];\n  }>;\n  headerExtensions: Partial<{\n    audio: RTCRtpHeaderExtensionParameters[];\n    video: RTCRtpHeaderExtensionParameters[];\n  }>;\n  iceTransportPolicy: \"all\" | \"relay\";\n  iceServers: RTCIceServer[];\n  /**Minimum port and Maximum port must not be the same value */\n  icePortRange: [number, number] | undefined;\n  iceInterfaceAddresses: InterfaceAddresses | undefined;\n  /** Add additional host (local) addresses to use for candidate gathering.\n   * Notably, you can include hosts that are normally excluded, such as loopback, tun interfaces, etc.\n   */\n  iceAdditionalHostAddresses: string[] | undefined;\n  iceUseIpv4: boolean;\n  iceUseIpv6: boolean;\n  forceTurnTCP: boolean;\n  /** If provided, is called on each STUN request.\n   * Return `true` if a STUN response should be sent, false if it should be skipped. */\n  iceFilterStunResponse:\n    | ((message: Message, addr: Address, protocol: Protocol) => boolean)\n    | undefined;\n  iceFilterCandidatePair: ((pair: CandidatePair) => boolean) | undefined;\n  dtls: Partial<{\n    keys: DtlsKeys;\n  }>;\n  icePasswordPrefix: string | undefined;\n  bundlePolicy: BundlePolicy;\n  debug: Partial<{\n    /**% */\n    inboundPacketLoss: number;\n    /**% */\n    outboundPacketLoss: number;\n    /**ms */\n    receiverReportDelay: number;\n    disableSendNack: boolean;\n    disableRecvRetransmit: boolean;\n  }>;\n  midSuffix: boolean;\n}\n\nexport const findCodecByMimeType = (\n  codecs: RTCRtpCodecParameters[],\n  target: RTCRtpCodecParameters,\n) =>\n  codecs.find(\n    (localCodec) =>\n      localCodec.mimeType.toLowerCase() === target.mimeType.toLowerCase(),\n  )\n    ? target\n    : undefined;\n\nexport type RTCIceServer = {\n  urls: string;\n  username?: string;\n  credential?: string;\n};\n\nexport const defaultPeerConfig: PeerConfig = {\n  codecs: {\n    audio: [\n      new RTCRtpCodecParameters({\n        mimeType: \"audio/opus\",\n        clockRate: 48000,\n        channels: 2,\n      }),\n      new RTCRtpCodecParameters({\n        mimeType: \"audio/PCMU\",\n        clockRate: 8000,\n        channels: 1,\n        payloadType: 0,\n      }),\n    ],\n    video: [\n      new RTCRtpCodecParameters({\n        mimeType: \"video/VP8\",\n        clockRate: 90000,\n        rtcpFeedback: [useNACK(), usePLI(), useREMB()],\n      }),\n    ],\n  },\n  headerExtensions: {\n    audio: [],\n    video: [],\n  },\n  iceTransportPolicy: \"all\",\n  iceServers: [{ urls: \"stun:stun.l.google.com:19302\" }],\n  icePortRange: undefined,\n  iceInterfaceAddresses: undefined,\n  iceAdditionalHostAddresses: undefined,\n  iceUseIpv4: true,\n  iceUseIpv6: true,\n  iceFilterStunResponse: undefined,\n  iceFilterCandidatePair: undefined,\n  icePasswordPrefix: undefined,\n  dtls: {},\n  bundlePolicy: \"max-compat\",\n  debug: {},\n  midSuffix: false,\n  forceTurnTCP: false,\n};\n\nexport interface RTCTrackEvent {\n  track: MediaStreamTrack;\n  streams: MediaStream[];\n  transceiver: RTCRtpTransceiver;\n  receiver: RTCRtpReceiver;\n}\n\nexport interface RTCDataChannelEvent {\n  channel: RTCDataChannel;\n}\n\nexport interface RTCPeerConnectionIceEvent {\n  candidate?: RTCIceCandidate;\n}\n\ntype Media = \"audio\" | \"video\";\n\nexport interface RTCSessionDescriptionInit {\n  sdp?: string;\n  type: RTCSdpType;\n}\nexport type RTCSdpType = \"answer\" | \"offer\" | \"pranswer\" | \"rollback\";\n"]}

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "werift",
-  "version": "0.21.0",
+  "version": "0.21.2",
   "description": "WebRTC Implementation for TypeScript (Node.js)",
   "keywords": [
     "WebRTC",
@@ -50,6 +50,7 @@
     "@noble/curves": "^1.3.0",
     "@peculiar/x509": "^1.9.2",
     "@shinyoshiaki/ebml-builder": "^0.0.1",
+    "multicast-dns": "^7.2.5",
     "aes-js": "^3.1.2",
     "@shinyoshiaki/binary-data": "^0.6.1",
     "buffer-crc32": "^1.0.0",

package/src/index.ts

@@ -17,3 +17,4 @@ export * from "./transport/ice";
 export * from "./transport/sctp";
 export * from "./types/domain";
 export * from "./utils";
+export * from "./media/codec";

package/src/media/codec.ts

@@ -0,0 +1,27 @@
+import { RTCRtpCodecParameters } from "..";
+
+export const useH264 = (props: Partial<RTCRtpCodecParameters> = {}) =>
+  new RTCRtpCodecParameters({
+    mimeType: "video/h264",
+    clockRate: 90000,
+    rtcpFeedback: [
+      { type: "nack" },
+      { type: "nack", parameter: "pli" },
+      { type: "goog-remb" },
+    ],
+    parameters:
+      "profile-level-id=42e01f;packetization-mode=1;level-asymmetry-allowed=1",
+    ...props,
+  });
+
+export const useVP8 = (props: Partial<RTCRtpCodecParameters> = {}) =>
+  new RTCRtpCodecParameters({
+    mimeType: "video/VP8",
+    clockRate: 90000,
+    rtcpFeedback: [
+      { type: "nack" },
+      { type: "nack", parameter: "pli" },
+      { type: "goog-remb" },
+    ],
+    ...props,
+  });

package/src/media/rtpReceiver.ts

@@ -36,7 +36,7 @@ import { NackHandler } from "./receiver/nack";
 import { ReceiverTWCC } from "./receiver/receiverTwcc";
 import { StreamStatistics } from "./receiver/statistics";
 
-import type { MediaStreamTrack } from "./track";
+import { MediaStreamTrack } from "./track";
 
 const log = debug("werift:packages/webrtc/src/media/rtpReceiver.ts");
 
@@ -91,7 +91,7 @@ export class RTCRtpReceiver {
 
   // todo fix
   get track() {
-    return this.tracks[0];
+    return this.tracks[0] ?? new MediaStreamTrack({ kind: this.kind });
   }
 
   get nackEnabled() {

package/src/nonstandard/navigator.ts

@@ -64,12 +64,14 @@ export class MediaDevices extends EventTarget {
   readonly getUdpMedia = ({
     port,
     codec,
+    kind,
   }: {
     port: number;
     codec: ConstructorParameters<typeof RTCRtpCodecParameters>[0];
+    kind: "audio" | "video";
   }) => {
     const track = new MediaStreamTrack({
-      kind: "audio",
+      kind,
       codec: new RTCRtpCodecParameters(codec),
     });
 
@@ -143,3 +145,5 @@ interface DoubleRange {
   max?: number;
   min?: number;
 }
+
+export const navigator = new Navigator();

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

@@ -1,5 +1,5 @@
 import { unlink } from "fs/promises";
-import { EventDisposer } from "../../../imports/common";
+import { Event, EventDisposer } from "../../../imports/common";
 
 import { MediaWriter } from ".";
 import { type MediaStreamTrack, WeriftError } from "../../..";
@@ -21,7 +21,8 @@ const sourcePath = "packages/webrtc/src/nonstandard/recorder/writer/webm.ts";
 
 export class WebmFactory extends MediaWriter {
   rtpSources: RtpSourceCallback[] = [];
-
+  private onEol = new Event();
+  private ended = false;
   unSubscribers = new EventDisposer();
 
   async start(tracks: MediaStreamTrack[]) {
@@ -140,7 +141,13 @@ export class WebmFactory extends MediaWriter {
       return rtpSource;
     });
     if (this.props.path) {
-      webm.pipe(saveToFileSystem(this.props.path));
+      webm.pipe(async (o) => {
+        const eol = await saveToFileSystem(this.props.path)(o);
+        if (eol) {
+          this.onEol.execute();
+          this.ended = true;
+        }
+      });
     } else if (this.props.stream) {
       webm.pipe(async (o) => {
         this.props.stream.execute(o);
@@ -150,6 +157,10 @@ export class WebmFactory extends MediaWriter {
 
   async stop() {
     await Promise.all(this.rtpSources.map((r) => r.stop()));
+
+    if (!this.ended) {
+      await this.onEol.asPromise(5000);
+    }
     this.unSubscribers.dispose();
   }
 }

package/src/peerConnection.ts

@@ -1300,7 +1300,9 @@ export class RTCPeerConnection extends EventTarget {
     /**todo impl */
     ms?: MediaStream,
   ) {
-    if (this.isClosed) throw new Error("is closed");
+    if (this.isClosed) {
+      throw new Error("is closed");
+    }
     if (this.getSenders().find((sender) => sender.track?.uuid === track.uuid)) {
       throw new Error("track exist");
     }