werift 0.14.2-debug2a → 0.14.4-debug0

This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
Files changed (34) hide show
  1. package/lib/dtls/src/transport.js +10 -1
  2. package/lib/dtls/src/transport.js.map +1 -1
  3. package/lib/ice/src/ice.js +1 -1
  4. package/lib/ice/src/ice.js.map +1 -1
  5. package/lib/ice/src/stun/protocol.js +8 -1
  6. package/lib/ice/src/stun/protocol.js.map +1 -1
  7. package/lib/ice/src/transport.js +9 -12
  8. package/lib/ice/src/transport.js.map +1 -1
  9. package/lib/rtp/src/rtp/red/encoder.js +6 -1
  10. package/lib/rtp/src/rtp/red/encoder.js.map +1 -1
  11. package/lib/rtp/src/srtp/cipher/ctr.js +1 -5
  12. package/lib/rtp/src/srtp/cipher/ctr.js.map +1 -1
  13. package/lib/sctp/src/helper.d.ts +1 -1
  14. package/lib/sctp/src/helper.js.map +1 -1
  15. package/lib/sctp/src/sctp.d.ts +6 -2
  16. package/lib/sctp/src/sctp.js +107 -41
  17. package/lib/sctp/src/sctp.js.map +1 -1
  18. package/lib/sctp/src/transport.js +8 -1
  19. package/lib/sctp/src/transport.js.map +1 -1
  20. package/lib/webrtc/src/media/rtpSender.d.ts +1 -1
  21. package/lib/webrtc/src/media/rtpSender.js +6 -6
  22. package/lib/webrtc/src/media/rtpSender.js.map +1 -1
  23. package/lib/webrtc/src/peerConnection.js +5 -1
  24. package/lib/webrtc/src/peerConnection.js.map +1 -1
  25. package/lib/webrtc/src/transport/dtls.d.ts +1 -1
  26. package/lib/webrtc/src/transport/dtls.js +7 -8
  27. package/lib/webrtc/src/transport/dtls.js.map +1 -1
  28. package/lib/webrtc/src/transport/sctp.js +30 -17
  29. package/lib/webrtc/src/transport/sctp.js.map +1 -1
  30. package/package.json +18 -20
  31. package/src/media/rtpSender.ts +9 -6
  32. package/src/peerConnection.ts +5 -1
  33. package/src/transport/dtls.ts +7 -7
  34. package/src/transport/sctp.ts +31 -24
package/lib/webrtc/src/transport/sctp.js.map CHANGED
@@ -1 +1 @@
1
- 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debug from \"debug\";\nimport { jspack } from \"jspack\";\nimport { Event } from \"rx.mini\";\nimport * as uuid from \"uuid\";\n\nimport { SCTP, SCTP_STATE, Transport } from \"../../../sctp/src\";\nimport {\n DATA_CHANNEL_ACK,\n DATA_CHANNEL_OPEN,\n DATA_CHANNEL_RELIABLE,\n WEBRTC_BINARY,\n WEBRTC_BINARY_EMPTY,\n WEBRTC_DCEP,\n WEBRTC_STRING,\n WEBRTC_STRING_EMPTY,\n} from \"../const\";\nimport {\n DCState,\n RTCDataChannel,\n RTCDataChannelParameters,\n} from \"../dataChannel\";\nimport { RTCDtlsTransport } from \"./dtls\";\n\nconst log = debug(\"werift/webrtc/transport/sctp\");\n\nexport class RTCSctpTransport {\n readonly onDataChannel = new Event<[RTCDataChannel]>();\n readonly uuid = uuid.v4();\n readonly sctp = new SCTP(new BridgeDtls(this.dtlsTransport), this.port);\n mid?: string;\n bundled = false;\n dataChannels: { [key: number]: RTCDataChannel } = {};\n\n private dataChannelQueue: [RTCDataChannel, number, Buffer][] = [];\n private dataChannelId?: number;\n\n constructor(public dtlsTransport: RTCDtlsTransport, public port = 5000) {\n this.sctp.onReceive.subscribe(this.datachannelReceive);\n this.sctp.onReconfigStreams.subscribe((ids: number[]) => {\n ids.forEach((id) => {\n const dc = this.dataChannels[id];\n if (!dc) return;\n // todo fix\n dc.setReadyState(\"closing\");\n dc.setReadyState(\"closed\");\n delete this.dataChannels[id];\n });\n });\n this.sctp.stateChanged.connected.subscribe(() => {\n Object.values(this.dataChannels).forEach((channel) => {\n if (channel.negotiated && channel.readyState !== \"open\") {\n channel.setReadyState(\"open\");\n }\n });\n this.dataChannelFlush();\n });\n this.sctp.stateChanged.closed.subscribe(() => {\n Object.values(this.dataChannels).forEach((dc) => {\n dc.setReadyState(\"closed\");\n });\n this.dataChannels = {};\n });\n this.sctp.onSackReceived = async () => {\n await this.dataChannelFlush();\n };\n\n this.dtlsTransport.onStateChange.subscribe((state) => {\n if (state === \"closed\") {\n this.sctp.setState(SCTP_STATE.CLOSED);\n }\n });\n }\n\n private get isServer() {\n return this.dtlsTransport.iceTransport.role !== \"controlling\";\n }\n\n channelByLabel(label: string) {\n return Object.values(this.dataChannels).find((d) => d.label === label);\n }\n\n private datachannelReceive = async (\n streamId: number,\n ppId: number,\n data: Buffer\n ) => {\n if (ppId === WEBRTC_DCEP && data.length > 0) {\n log(\"DCEP\", streamId, ppId, data);\n switch (data[0]) {\n case DATA_CHANNEL_OPEN:\n {\n if (data.length >= 12) {\n if (Object.keys(this.dataChannels).includes(streamId.toString()))\n throw new Error();\n\n const [\n ,\n channelType,\n ,\n reliability,\n labelLength,\n protocolLength,\n ] = jspack.Unpack(\"!BBHLHH\", data);\n\n let pos = 12;\n const label = data.slice(pos, pos + labelLength).toString(\"utf8\");\n pos += labelLength;\n const protocol = data\n .slice(pos, pos + protocolLength)\n .toString(\"utf8\");\n\n log(\"DATA_CHANNEL_OPEN\", {\n channelType,\n reliability,\n streamId,\n label,\n protocol,\n });\n\n const maxRetransmits =\n (channelType & 0x03) === 1 ? reliability : undefined;\n const maxPacketLifeTime =\n (channelType & 0x03) === 2 ? reliability : undefined;\n\n // # register channel\n const parameters = new RTCDataChannelParameters({\n label,\n ordered: (channelType & 0x80) === 0,\n maxPacketLifeTime,\n maxRetransmits,\n protocol,\n id: streamId,\n });\n const channel = new RTCDataChannel(this, parameters, false);\n channel.isCreatedByRemote = true;\n this.dataChannels[streamId] = channel;\n\n this.dataChannelQueue.push([\n channel,\n WEBRTC_DCEP,\n Buffer.from(jspack.Pack(\"!B\", [DATA_CHANNEL_ACK])),\n ]);\n\n this.onDataChannel.execute(channel);\n channel.setReadyState(\"open\");\n\n await this.dataChannelFlush();\n }\n }\n break;\n case DATA_CHANNEL_ACK:\n log(\"DATA_CHANNEL_ACK\");\n const channel = this.dataChannels[streamId];\n if (!channel) throw new Error();\n channel.setReadyState(\"open\");\n break;\n }\n } else {\n const channel = this.dataChannels[streamId];\n if (channel) {\n const msg = (() => {\n switch (ppId) {\n case WEBRTC_STRING:\n return data.toString(\"utf8\");\n case WEBRTC_STRING_EMPTY:\n return \"\";\n case WEBRTC_BINARY:\n return data;\n case WEBRTC_BINARY_EMPTY:\n return Buffer.from([]);\n default:\n throw new Error();\n }\n })();\n channel.message.execute(msg);\n channel.emit(\"message\", { data: msg });\n if (channel.onmessage) channel.onmessage({ data: msg });\n }\n }\n };\n\n dataChannelAddNegotiated(channel: RTCDataChannel) {\n if (channel.id == undefined) {\n throw new Error();\n }\n if (this.dataChannels[channel.id]) {\n throw new Error();\n }\n\n this.dataChannels[channel.id] = channel;\n\n if (this.sctp.associationState === SCTP_STATE.ESTABLISHED) {\n channel.setReadyState(\"open\");\n }\n }\n\n dataChannelOpen(channel: RTCDataChannel) {\n if (channel.id) {\n if (this.dataChannels[channel.id])\n throw new Error(\n `Data channel with ID ${channel.id} already registered`\n );\n this.dataChannels[channel.id] = channel;\n }\n\n let channelType = DATA_CHANNEL_RELIABLE;\n const priority = 0;\n let reliability = 0;\n\n if (!channel.ordered) {\n channelType = 0x80;\n }\n if (channel.maxRetransmits) {\n channelType = 1;\n reliability = channel.maxRetransmits;\n } else if (channel.maxPacketLifeTime) {\n channelType = 2;\n reliability = channel.maxPacketLifeTime;\n }\n\n // 5.1. DATA_CHANNEL_OPEN Message\n const data = jspack.Pack(\"!BBHLHH\", [\n DATA_CHANNEL_OPEN,\n channelType,\n priority,\n reliability,\n channel.label.length,\n channel.protocol.length,\n ]);\n const send = Buffer.concat([\n Buffer.from(data),\n Buffer.from(channel.label, \"utf8\"),\n Buffer.from(channel.protocol, \"utf8\"),\n ]);\n this.dataChannelQueue.push([channel, WEBRTC_DCEP, send]);\n this.dataChannelFlush();\n }\n\n private async dataChannelFlush() {\n // \"\"\"\n // Try to flush buffered data to the SCTP layer.\n\n // We wait until the association is established, as we need to know\n // whether we are a client or a server to correctly assign an odd/even ID\n // to the data channels.\n // \"\"\"\n\n if (this.sctp.associationState != SCTP_STATE.ESTABLISHED) return;\n if (this.sctp.outboundQueue.length > 0) return;\n\n while (this.dataChannelQueue.length > 0) {\n const [channel, protocol, userData] = this.dataChannelQueue.shift()!;\n\n let streamId = channel.id;\n if (streamId === undefined) {\n streamId = this.dataChannelId!;\n while (Object.keys(this.dataChannels).includes(streamId.toString())) {\n streamId += 2;\n }\n this.dataChannels[streamId] = channel;\n channel.setId(streamId);\n }\n\n if (protocol === WEBRTC_DCEP) {\n await this.sctp.send(streamId, protocol, userData);\n } else {\n const expiry = channel.maxPacketLifeTime\n ? Date.now() + channel.maxPacketLifeTime / 1000\n : undefined;\n\n await this.sctp.send(\n streamId,\n protocol,\n userData,\n expiry,\n channel.maxRetransmits,\n channel.ordered\n );\n channel.addBufferedAmount(-userData.length);\n }\n }\n }\n\n datachannelSend = (channel: RTCDataChannel, data: Buffer | string) => {\n channel.addBufferedAmount(data.length);\n\n this.dataChannelQueue.push(\n typeof data === \"string\"\n ? [channel, WEBRTC_STRING, Buffer.from(data)]\n : [channel, WEBRTC_BINARY, data]\n );\n\n if (this.sctp.associationState !== SCTP_STATE.ESTABLISHED) {\n log(\"sctp not established\", this.sctp.associationState);\n }\n\n this.dataChannelFlush();\n };\n\n static getCapabilities() {\n return new RTCSctpCapabilities(65536);\n }\n\n setRemotePort(port: number) {\n this.sctp.setRemotePort(port);\n }\n\n async start(remotePort: number) {\n if (this.isServer) {\n this.dataChannelId = 0;\n } else {\n this.dataChannelId = 1;\n }\n this.sctp.isServer = this.isServer;\n\n this.setRemotePort(remotePort);\n await this.sctp.start();\n }\n\n async stop() {\n this.dtlsTransport.dataReceiver = () => {};\n await this.sctp.stop();\n }\n\n dataChannelClose(channel: RTCDataChannel) {\n if (!([\"closing\", \"closed\"] as DCState[]).includes(channel.readyState)) {\n channel.setReadyState(\"closing\");\n\n if (this.sctp.associationState === SCTP_STATE.ESTABLISHED) {\n this.sctp.reconfigQueue.push(channel.id);\n if (this.sctp.reconfigQueue.length === 1) {\n this.sctp.transmitReconfigRequest();\n }\n } else {\n this.dataChannelQueue = this.dataChannelQueue.filter(\n (queueItem) => queueItem[0].id !== channel.id\n );\n if (channel.id) {\n delete this.dataChannels[channel.id];\n }\n channel.setReadyState(\"closed\");\n }\n }\n }\n}\n\nexport class RTCSctpCapabilities {\n constructor(public maxMessageSize: number) {}\n}\n\nclass BridgeDtls implements Transport {\n constructor(private dtls: RTCDtlsTransport) {}\n set onData(onData: (buf: Buffer) => void) {\n this.dtls.dataReceiver = onData;\n }\n readonly send = this.dtls.sendData;\n close() {}\n}\n"]}
1
+ 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debug from \"debug\";\nimport { jspack } from \"jspack\";\nimport { Event } from \"rx.mini\";\nimport * as uuid from \"uuid\";\n\nimport { SCTP, SCTP_STATE, Transport } from \"../../../sctp/src\";\nimport {\n DATA_CHANNEL_ACK,\n DATA_CHANNEL_OPEN,\n DATA_CHANNEL_RELIABLE,\n WEBRTC_BINARY,\n WEBRTC_BINARY_EMPTY,\n WEBRTC_DCEP,\n WEBRTC_STRING,\n WEBRTC_STRING_EMPTY,\n} from \"../const\";\nimport {\n DCState,\n RTCDataChannel,\n RTCDataChannelParameters,\n} from \"../dataChannel\";\nimport { RTCDtlsTransport } from \"./dtls\";\n\nconst log = debug(\"werift/webrtc/transport/sctp\");\n\nexport class RTCSctpTransport {\n readonly onDataChannel = new Event<[RTCDataChannel]>();\n readonly uuid = uuid.v4();\n readonly sctp = new SCTP(new BridgeDtls(this.dtlsTransport), this.port);\n mid?: string;\n bundled = false;\n dataChannels: { [key: number]: RTCDataChannel } = {};\n\n private dataChannelQueue: [RTCDataChannel, number, Buffer][] = [];\n private dataChannelId?: number;\n\n constructor(public dtlsTransport: RTCDtlsTransport, public port = 5000) {\n this.sctp.onReceive.subscribe(this.datachannelReceive);\n this.sctp.onReconfigStreams.subscribe((ids: number[]) => {\n ids.forEach((id) => {\n const dc = this.dataChannels[id];\n if (!dc) return;\n // todo fix\n dc.setReadyState(\"closing\");\n dc.setReadyState(\"closed\");\n delete this.dataChannels[id];\n });\n });\n this.sctp.stateChanged.connected.subscribe(() => {\n Object.values(this.dataChannels).forEach((channel) => {\n if (channel.negotiated && channel.readyState !== \"open\") {\n channel.setReadyState(\"open\");\n }\n });\n this.dataChannelFlush();\n });\n this.sctp.stateChanged.closed.subscribe(() => {\n Object.values(this.dataChannels).forEach((dc) => {\n dc.setReadyState(\"closed\");\n });\n this.dataChannels = {};\n });\n this.sctp.onSackReceived = async () => {\n await this.dataChannelFlush();\n };\n\n this.dtlsTransport.onStateChange.subscribe((state) => {\n if (state === \"closed\") {\n this.sctp.setState(SCTP_STATE.CLOSED);\n }\n });\n }\n\n private get isServer() {\n return this.dtlsTransport.iceTransport.role !== \"controlling\";\n }\n\n channelByLabel(label: string) {\n return Object.values(this.dataChannels).find((d) => d.label === label);\n }\n\n private datachannelReceive = async (\n streamId: number,\n ppId: number,\n data: Buffer\n ) => {\n if (ppId === WEBRTC_DCEP && data.length > 0) {\n log(\"DCEP\", streamId, ppId, data);\n switch (data[0]) {\n case DATA_CHANNEL_OPEN:\n {\n if (data.length < 12) {\n log(\"DATA_CHANNEL_OPEN data.length not enough\");\n return;\n }\n\n if (!Object.keys(this.dataChannels).includes(streamId.toString())) {\n const [\n ,\n channelType,\n ,\n reliability,\n labelLength,\n protocolLength,\n ] = jspack.Unpack(\"!BBHLHH\", data);\n\n let pos = 12;\n const label = data.slice(pos, pos + labelLength).toString(\"utf8\");\n pos += labelLength;\n const protocol = data\n .slice(pos, pos + protocolLength)\n .toString(\"utf8\");\n\n log(\"DATA_CHANNEL_OPEN\", {\n channelType,\n reliability,\n streamId,\n label,\n protocol,\n });\n\n const maxRetransmits =\n (channelType & 0x03) === 1 ? reliability : undefined;\n const maxPacketLifeTime =\n (channelType & 0x03) === 2 ? reliability : undefined;\n\n // # register channel\n const parameters = new RTCDataChannelParameters({\n label,\n ordered: (channelType & 0x80) === 0,\n maxPacketLifeTime,\n maxRetransmits,\n protocol,\n id: streamId,\n });\n const channel = new RTCDataChannel(this, parameters, false);\n channel.isCreatedByRemote = true;\n this.dataChannels[streamId] = channel;\n\n this.onDataChannel.execute(channel);\n channel.setReadyState(\"open\");\n } else {\n log(\"datachannel already opened\", \"retransmit ack\");\n }\n\n const channel = this.dataChannels[streamId];\n this.dataChannelQueue.push([\n channel,\n WEBRTC_DCEP,\n Buffer.from(jspack.Pack(\"!B\", [DATA_CHANNEL_ACK])),\n ]);\n await this.dataChannelFlush();\n }\n break;\n case DATA_CHANNEL_ACK:\n log(\"DATA_CHANNEL_ACK\", streamId, ppId);\n const channel = this.dataChannels[streamId];\n if (!channel) {\n throw new Error(\"channel not found\");\n }\n channel.setReadyState(\"open\");\n break;\n }\n } else {\n const channel = this.dataChannels[streamId];\n if (channel) {\n const msg = (() => {\n switch (ppId) {\n case WEBRTC_STRING:\n return data.toString(\"utf8\");\n case WEBRTC_STRING_EMPTY:\n return \"\";\n case WEBRTC_BINARY:\n return data;\n case WEBRTC_BINARY_EMPTY:\n return Buffer.from([]);\n default:\n throw new Error();\n }\n })();\n\n channel.message.execute(msg);\n channel.emit(\"message\", { data: msg });\n if (channel.onmessage) {\n channel.onmessage({ data: msg });\n }\n }\n }\n };\n\n dataChannelAddNegotiated(channel: RTCDataChannel) {\n if (channel.id == undefined) {\n throw new Error();\n }\n if (this.dataChannels[channel.id]) {\n throw new Error();\n }\n\n this.dataChannels[channel.id] = channel;\n\n if (this.sctp.associationState === SCTP_STATE.ESTABLISHED) {\n channel.setReadyState(\"open\");\n }\n }\n\n dataChannelOpen(channel: RTCDataChannel) {\n if (channel.id) {\n if (this.dataChannels[channel.id])\n throw new Error(\n `Data channel with ID ${channel.id} already registered`\n );\n this.dataChannels[channel.id] = channel;\n }\n\n let channelType = DATA_CHANNEL_RELIABLE;\n const priority = 0;\n let reliability = 0;\n\n if (!channel.ordered) {\n channelType = 0x80;\n }\n if (channel.maxRetransmits) {\n channelType = 1;\n reliability = channel.maxRetransmits;\n } else if (channel.maxPacketLifeTime) {\n channelType = 2;\n reliability = channel.maxPacketLifeTime;\n }\n\n // 5.1. DATA_CHANNEL_OPEN Message\n const data = jspack.Pack(\"!BBHLHH\", [\n DATA_CHANNEL_OPEN,\n channelType,\n priority,\n reliability,\n channel.label.length,\n channel.protocol.length,\n ]);\n const send = Buffer.concat([\n Buffer.from(data),\n Buffer.from(channel.label, \"utf8\"),\n Buffer.from(channel.protocol, \"utf8\"),\n ]);\n this.dataChannelQueue.push([channel, WEBRTC_DCEP, send]);\n this.dataChannelFlush();\n }\n\n private async dataChannelFlush() {\n // \"\"\"\n // Try to flush buffered data to the SCTP layer.\n\n // We wait until the association is established, as we need to know\n // whether we are a client or a server to correctly assign an odd/even ID\n // to the data channels.\n // \"\"\"\n\n if (this.sctp.associationState != SCTP_STATE.ESTABLISHED) return;\n if (this.sctp.outboundQueue.length > 0) return;\n\n while (this.dataChannelQueue.length > 0) {\n const [channel, protocol, userData] = this.dataChannelQueue.shift()!;\n\n let streamId = channel.id;\n if (streamId === undefined) {\n streamId = this.dataChannelId!;\n while (Object.keys(this.dataChannels).includes(streamId.toString())) {\n streamId += 2;\n }\n this.dataChannels[streamId] = channel;\n channel.setId(streamId);\n }\n\n if (protocol === WEBRTC_DCEP) {\n await this.sctp.send(streamId, protocol, userData, {\n ordered: true,\n });\n } else {\n const expiry = channel.maxPacketLifeTime\n ? Date.now() + channel.maxPacketLifeTime / 1000\n : undefined;\n\n await this.sctp.send(streamId, protocol, userData, {\n expiry,\n maxRetransmits: channel.maxRetransmits,\n ordered: channel.ordered,\n });\n channel.addBufferedAmount(-userData.length);\n }\n }\n }\n\n datachannelSend = (channel: RTCDataChannel, data: Buffer | string) => {\n channel.addBufferedAmount(data.length);\n\n this.dataChannelQueue.push(\n typeof data === \"string\"\n ? [channel, WEBRTC_STRING, Buffer.from(data)]\n : [channel, WEBRTC_BINARY, data]\n );\n\n if (this.sctp.associationState !== SCTP_STATE.ESTABLISHED) {\n log(\"sctp not established\", this.sctp.associationState);\n }\n\n this.dataChannelFlush();\n };\n\n static getCapabilities() {\n return new RTCSctpCapabilities(65536);\n }\n\n setRemotePort(port: number) {\n this.sctp.setRemotePort(port);\n }\n\n async start(remotePort: number) {\n if (this.isServer) {\n this.dataChannelId = 0;\n } else {\n this.dataChannelId = 1;\n }\n this.sctp.isServer = this.isServer;\n\n await this.sctp.start(remotePort);\n }\n\n async stop() {\n this.dtlsTransport.dataReceiver = () => {};\n await this.sctp.stop();\n }\n\n dataChannelClose(channel: RTCDataChannel) {\n if (!([\"closing\", \"closed\"] as DCState[]).includes(channel.readyState)) {\n channel.setReadyState(\"closing\");\n\n if (this.sctp.associationState === SCTP_STATE.ESTABLISHED) {\n this.sctp.reconfigQueue.push(channel.id);\n if (this.sctp.reconfigQueue.length === 1) {\n this.sctp.transmitReconfigRequest();\n }\n } else {\n this.dataChannelQueue = this.dataChannelQueue.filter(\n (queueItem) => queueItem[0].id !== channel.id\n );\n if (channel.id) {\n delete this.dataChannels[channel.id];\n }\n channel.setReadyState(\"closed\");\n }\n }\n }\n}\n\nexport class RTCSctpCapabilities {\n constructor(public maxMessageSize: number) {}\n}\n\nclass BridgeDtls implements Transport {\n constructor(private dtls: RTCDtlsTransport) {}\n set onData(onData: (buf: Buffer) => void) {\n this.dtls.dataReceiver = onData;\n }\n readonly send = this.dtls.sendData;\n close() {}\n}\n"]}
package/package.json CHANGED
@@ -1,6 +1,6 @@
1
1
  {
2
2
  "name": "werift",
3
- "version": "0.14.2-debug2a",
3
+ "version": "0.14.4-debug0",
4
4
  "description": "WebRTC Implementation for TypeScript (Node.js)",
5
5
  "keywords": [
6
6
  "WebRTC",
@@ -24,11 +24,11 @@
24
24
  "scripts": {
25
25
  "build": "rm -rf lib && yarn format && tsc -p ./tsconfig.production.json",
26
26
  "depcheck": "npx depcheck --ignores='@typescript-eslint/*,eslint-*'",
27
+ "deploy": "yarn build && yarn publish",
27
28
  "doc": "rm -rf doc && typedoc --out doc ./src/index.ts",
28
29
  "format": "yarn unused && yarn lint && prettier --write src/**/*.ts",
29
30
  "lint": "eslint ./src --fix",
30
31
  "publish:tag": "yarn publish --tag maintain-dtls",
31
- "deploy": "yarn build && yarn publish",
32
32
  "test": "npm run type && jest --runInBand --forceExit --coverage",
33
33
  "type": "tsc --noEmit --project ./tsconfig.json",
34
34
  "unused": "organize-imports-cli src/**/*.ts"
@@ -40,11 +40,10 @@
40
40
  "@peculiar/x509": "^1.2.2",
41
41
  "@shinyoshiaki/ebml-builder": "^0.0.1",
42
42
  "aes-js": "^3.1.2",
43
- "big-integer": "^1.6.49",
44
43
  "binary-data": "^0.6.0",
45
44
  "buffer-crc32": "^0.2.13",
46
- "date-fns": "^2.24.0",
47
- "debug": "^4.3.1",
45
+ "date-fns": "^2.27.0",
46
+ "debug": "^4.3.3",
48
47
  "elliptic": "^6.5.3",
49
48
  "int64-buffer": "^1.0.1",
50
49
  "ip": "^1.1.5",
@@ -59,25 +58,24 @@
59
58
  },
60
59
  "devDependencies": {
61
60
  "@types/aes-js": "^3.1.1",
62
- "@types/big-integer": "^0.0.31",
63
61
  "@types/buffer-crc32": "^0.2.0",
64
62
  "@types/debug": "^4.1.7",
65
- "@types/elliptic": "^6.4.12",
63
+ "@types/elliptic": "^6.4.14",
66
64
  "@types/ip": "^1.1.0",
67
- "@types/jest": "^27.0.2",
68
- "@types/lodash": "^4.14.175",
69
- "@types/node": "^16.10.2",
70
- "@types/uuid": "^8.3.0",
71
- "@typescript-eslint/eslint-plugin": "^4.32.0",
72
- "@typescript-eslint/parser": "^4.32.0",
65
+ "@types/jest": "^27.0.3",
66
+ "@types/lodash": "^4.14.178",
67
+ "@types/node": "^17.0.0",
68
+ "@types/uuid": "^8.3.3",
69
+ "@typescript-eslint/eslint-plugin": "^5.7.0",
70
+ "@typescript-eslint/parser": "^5.7.0",
73
71
  "eslint-plugin-prettier": "^4.0.0",
74
- "jest": "^27.2.4",
75
- "node-actionlint": "^1.2.0",
76
- "prettier": "^2.4.1",
77
- "ts-jest": "^27.0.5",
78
- "ts-node": "^10.2.1",
79
- "typedoc": "^0.22.4",
80
- "typescript": "^4.4.3"
72
+ "jest": "^27.4.5",
73
+ "node-actionlint": "^1.2.1",
74
+ "prettier": "^2.5.1",
75
+ "ts-jest": "^27.1.1",
76
+ "ts-node": "^10.4.0",
77
+ "typedoc": "^0.22.10",
78
+ "typescript": "^4.5.4"
81
79
  },
82
80
  "engines": {
83
81
  "node": ">=15"
package/src/media/rtpSender.ts CHANGED
@@ -161,8 +161,8 @@ export class RTCRtpSender {
161
161
 
162
162
  track.id = this.trackId;
163
163
 
164
- const { unSubscribe } = track.onReceiveRtp.subscribe((rtp) => {
165
- this.sendRtp(rtp);
164
+ const { unSubscribe } = track.onReceiveRtp.subscribe(async (rtp) => {
165
+ await this.sendRtp(rtp);
166
166
  });
167
167
  this.track = track;
168
168
  this.disposeTrack = unSubscribe;
@@ -266,7 +266,7 @@ export class RTCRtpSender {
266
266
  log("replaceRTP", this.sequenceNumber, sequenceNumber, this.seqOffset);
267
267
  }
268
268
 
269
- sendRtp(rtp: Buffer | RtpPacket) {
269
+ async sendRtp(rtp: Buffer | RtpPacket) {
270
270
  if (this.dtlsTransport.state !== "connected" || !this.codec) {
271
271
  return;
272
272
  }
@@ -345,7 +345,7 @@ export class RTCRtpSender {
345
345
  rtpPayload = red.serialize();
346
346
  }
347
347
 
348
- const size = this.dtlsTransport.sendRtp(rtpPayload, header);
348
+ const size = await this.dtlsTransport.sendRtp(rtpPayload, header);
349
349
 
350
350
  this.runRtcp();
351
351
  const sentInfo: SentInfo = {
@@ -391,7 +391,7 @@ export class RTCRtpSender {
391
391
  case GenericNack.count:
392
392
  {
393
393
  const feedback = packet.feedback as GenericNack;
394
- feedback.lost.forEach((seqNum) => {
394
+ feedback.lost.forEach(async (seqNum) => {
395
395
  let packet = this.rtpCache.find(
396
396
  (rtp) => rtp.header.sequenceNumber === seqNum
397
397
  );
@@ -408,7 +408,10 @@ export class RTCRtpSender {
408
408
  1
409
409
  );
410
410
  }
411
- this.dtlsTransport.sendRtp(packet.payload, packet.header);
411
+ await this.dtlsTransport.sendRtp(
412
+ packet.payload,
413
+ packet.header
414
+ );
412
415
  }
413
416
  });
414
417
  this.onGenericNack.execute(feedback);
package/src/peerConnection.ts CHANGED
@@ -318,8 +318,9 @@ export class RTCPeerConnection extends EventTarget {
318
318
  };
319
319
  const settings: Required<typeof base> = { ...base, ...options } as any;
320
320
 
321
- if (settings.maxPacketLifeTime && settings.maxRetransmits)
321
+ if (settings.maxPacketLifeTime && settings.maxRetransmits) {
322
322
  throw new Error("can not select both");
323
+ }
323
324
 
324
325
  if (!this.sctpTransport) {
325
326
  this.sctpTransport = this.createSctpTransport();
@@ -514,6 +515,7 @@ export class RTCPeerConnection extends EventTarget {
514
515
 
515
516
  // connect transports
516
517
  if (description.type === "answer") {
518
+ log("callee start connect");
517
519
  this.connect().catch((err) => {
518
520
  log("connect failed", err);
519
521
  this.setConnectionState("failed");
@@ -564,6 +566,7 @@ export class RTCPeerConnection extends EventTarget {
564
566
  await this.sctpTransport.start(this.sctpRemotePort);
565
567
  await this.sctpTransport.sctp.stateChanged.connected.asPromise();
566
568
  }
569
+ log("sctp connected");
567
570
 
568
571
  this.masterTransportEstablished = true;
569
572
  this.setConnectionState("connected");
@@ -781,6 +784,7 @@ export class RTCPeerConnection extends EventTarget {
781
784
 
782
785
  // connect transports
783
786
  if (remoteSdp.type === "answer") {
787
+ log("caller start connect");
784
788
  this.connect().catch((err) => {
785
789
  log("connect failed", err);
786
790
  this.setConnectionState("failed");
package/src/transport/dtls.ts CHANGED
@@ -194,20 +194,20 @@ export class RTCDtlsTransport {
194
194
  await this.dtls.send(data);
195
195
  };
196
196
 
197
- sendRtp(payload: Buffer, header: RtpHeader) {
197
+ async sendRtp(payload: Buffer, header: RtpHeader) {
198
198
  const enc = this.srtp.encrypt(payload, header);
199
- this.iceTransport.connection.send(enc);
199
+ await this.iceTransport.connection.send(enc).catch((err: Error) => {
200
+ log("sendRtp failed", err.message);
201
+ });
200
202
  return enc.length;
201
203
  }
202
204
 
203
205
  async sendRtcp(packets: RtcpPacket[]) {
204
206
  const payload = Buffer.concat(packets.map((packet) => packet.serialize()));
205
207
  const enc = this.srtcp.encrypt(payload);
206
- try {
207
- await this.iceTransport.connection.send(enc);
208
- } catch (error) {
209
- throw new Error("ice");
210
- }
208
+ await this.iceTransport.connection.send(enc).catch((err: Error) => {
209
+ log("sendRtcp failed", err.message);
210
+ });
211
211
  }
212
212
 
213
213
  private setState(state: DtlsState) {
package/src/transport/sctp.ts CHANGED
@@ -89,10 +89,12 @@ export class RTCSctpTransport {
89
89
  switch (data[0]) {
90
90
  case DATA_CHANNEL_OPEN:
91
91
  {
92
- if (data.length >= 12) {
93
- if (Object.keys(this.dataChannels).includes(streamId.toString()))
94
- throw new Error();
92
+ if (data.length < 12) {
93
+ log("DATA_CHANNEL_OPEN data.length not enough");
94
+ return;
95
+ }
95
96
 
97
+ if (!Object.keys(this.dataChannels).includes(streamId.toString())) {
96
98
  const [
97
99
  ,
98
100
  channelType,
@@ -135,23 +137,27 @@ export class RTCSctpTransport {
135
137
  channel.isCreatedByRemote = true;
136
138
  this.dataChannels[streamId] = channel;
137
139
 
138
- this.dataChannelQueue.push([
139
- channel,
140
- WEBRTC_DCEP,
141
- Buffer.from(jspack.Pack("!B", [DATA_CHANNEL_ACK])),
142
- ]);
143
-
144
140
  this.onDataChannel.execute(channel);
145
141
  channel.setReadyState("open");
146
-
147
- await this.dataChannelFlush();
142
+ } else {
143
+ log("datachannel already opened", "retransmit ack");
148
144
  }
145
+
146
+ const channel = this.dataChannels[streamId];
147
+ this.dataChannelQueue.push([
148
+ channel,
149
+ WEBRTC_DCEP,
150
+ Buffer.from(jspack.Pack("!B", [DATA_CHANNEL_ACK])),
151
+ ]);
152
+ await this.dataChannelFlush();
149
153
  }
150
154
  break;
151
155
  case DATA_CHANNEL_ACK:
152
- log("DATA_CHANNEL_ACK");
156
+ log("DATA_CHANNEL_ACK", streamId, ppId);
153
157
  const channel = this.dataChannels[streamId];
154
- if (!channel) throw new Error();
158
+ if (!channel) {
159
+ throw new Error("channel not found");
160
+ }
155
161
  channel.setReadyState("open");
156
162
  break;
157
163
  }
@@ -172,9 +178,12 @@ export class RTCSctpTransport {
172
178
  throw new Error();
173
179
  }
174
180
  })();
181
+
175
182
  channel.message.execute(msg);
176
183
  channel.emit("message", { data: msg });
177
- if (channel.onmessage) channel.onmessage({ data: msg });
184
+ if (channel.onmessage) {
185
+ channel.onmessage({ data: msg });
186
+ }
178
187
  }
179
188
  }
180
189
  };
@@ -262,20 +271,19 @@ export class RTCSctpTransport {
262
271
  }
263
272
 
264
273
  if (protocol === WEBRTC_DCEP) {
265
- await this.sctp.send(streamId, protocol, userData);
274
+ await this.sctp.send(streamId, protocol, userData, {
275
+ ordered: true,
276
+ });
266
277
  } else {
267
278
  const expiry = channel.maxPacketLifeTime
268
279
  ? Date.now() + channel.maxPacketLifeTime / 1000
269
280
  : undefined;
270
281
 
271
- await this.sctp.send(
272
- streamId,
273
- protocol,
274
- userData,
282
+ await this.sctp.send(streamId, protocol, userData, {
275
283
  expiry,
276
- channel.maxRetransmits,
277
- channel.ordered
278
- );
284
+ maxRetransmits: channel.maxRetransmits,
285
+ ordered: channel.ordered,
286
+ });
279
287
  channel.addBufferedAmount(-userData.length);
280
288
  }
281
289
  }
@@ -313,8 +321,7 @@ export class RTCSctpTransport {
313
321
  }
314
322
  this.sctp.isServer = this.isServer;
315
323
 
316
- this.setRemotePort(remotePort);
317
- await this.sctp.start();
324
+ await this.sctp.start(remotePort);
318
325
  }
319
326
 
320
327
  async stop() {