From 1569fec0cf23442b176b2a9a775208349aa719ed Mon Sep 17 00:00:00 2001 From: Madeline <46743919+MaddyUnderStars@users.noreply.github.com> Date: Sun, 16 Jan 2022 02:37:38 +1100 Subject: boilerplate stuff --- webrtc/src/start.ts | 1 + 1 file changed, 1 insertion(+) (limited to 'webrtc/src/start.ts') diff --git a/webrtc/src/start.ts b/webrtc/src/start.ts index 68867a2c..5614982d 100644 --- a/webrtc/src/start.ts +++ b/webrtc/src/start.ts @@ -1,3 +1,4 @@ import { Server } from "./Server"; const server = new Server(); +server.listen(); \ No newline at end of file -- cgit 1.5.1 From 7c87f8d60a15359a129ab076c212cbf4f1cd8884 Mon Sep 17 00:00:00 2001 From: Madeline <46743919+MaddyUnderStars@users.noreply.github.com> Date: Mon, 17 Jan 2022 02:59:26 +1100 Subject: more fuckery --- webrtc/package-lock.json | 14 ++++++++ webrtc/package.json | 1 + webrtc/src/Server.ts | 31 ++++++++++------ webrtc/src/opcodes/Connect.ts | 4 +++ webrtc/src/opcodes/Identify.ts | 68 ++++++++++++++++++++++++++++++++--- webrtc/src/opcodes/SelectProtocol.ts | 70 +++++++++++++++++++++++++++++++++--- webrtc/src/start.ts | 6 ++++ 7 files changed, 175 insertions(+), 19 deletions(-) (limited to 'webrtc/src/start.ts') diff --git a/webrtc/package-lock.json b/webrtc/package-lock.json index 6c3726dc..43bb8cd8 100644 --- a/webrtc/package-lock.json +++ b/webrtc/package-lock.json @@ -9,6 +9,7 @@ "version": "1.0.0", "license": "ISC", "dependencies": { + "dotenv": "^12.0.4", "mediasoup": "^3.9.5", "node-turn": "^0.0.6", "tsconfig-paths": "^3.12.0", @@ -210,6 +211,14 @@ "node": ">=0.3.1" } }, + "node_modules/dotenv": { + "version": "12.0.4", + "resolved": "https://registry.npmjs.org/dotenv/-/dotenv-12.0.4.tgz", + "integrity": "sha512-oWdqbSywffzH1l4WXKPHWA0TWYpqp7IyLfqjipT4upoIFS0HPMqtNotykQpD4iIg0BqtNmdgPCh2WMvMt7yTiw==", + "engines": { + "node": ">=12" + } + }, "node_modules/esprima": { "version": "4.0.1", "resolved": "https://registry.npmjs.org/esprima/-/esprima-4.0.1.tgz", @@ -672,6 +681,11 @@ "integrity": "sha512-58lmxKSA4BNyLz+HHMUzlOEpg09FV+ev6ZMe3vJihgdxzgcwZ8VoEEPmALCZG9LmqfVoNMMKpttIYTVG6uDY7A==", "dev": true }, + "dotenv": { + "version": "12.0.4", + "resolved": "https://registry.npmjs.org/dotenv/-/dotenv-12.0.4.tgz", + "integrity": "sha512-oWdqbSywffzH1l4WXKPHWA0TWYpqp7IyLfqjipT4upoIFS0HPMqtNotykQpD4iIg0BqtNmdgPCh2WMvMt7yTiw==" + }, "esprima": { "version": "4.0.1", "resolved": "https://registry.npmjs.org/esprima/-/esprima-4.0.1.tgz", diff --git a/webrtc/package.json b/webrtc/package.json index 8c66245d..d5a994a1 100644 --- a/webrtc/package.json +++ b/webrtc/package.json @@ -18,6 +18,7 @@ "typescript": "^4.3.2" }, "dependencies": { + "dotenv": "^12.0.4", "mediasoup": "^3.9.5", "node-turn": "^0.0.6", "tsconfig-paths": "^3.12.0", diff --git a/webrtc/src/Server.ts b/webrtc/src/Server.ts index cdda10ec..1d2e73e7 100644 --- a/webrtc/src/Server.ts +++ b/webrtc/src/Server.ts @@ -54,21 +54,32 @@ export class Server { }); worker.observer.on("newrouter", async (router: MediasoupTypes.Router) => { - console.log("new router"); + console.log("new router created [id:%s]", router.id); this.mediasoupRouters.push(router); - router.observer.on("newtransport", (transport: MediasoupTypes.Transport) => { - console.log("new transport"); + router.observer.on("newtransport", async (transport: MediasoupTypes.Transport) => { + console.log("new transport created [id:%s]", transport.id); - this.mediasoupTransports.push(transport); - }) + await transport.enableTraceEvent(); + + transport.observer.on("newproducer", (producer: MediasoupTypes.Producer) => { + console.log("new producer created [id:%s]", producer.id); + }); + + transport.observer.on("newconsumer", (consumer: MediasoupTypes.Consumer) => { + console.log("new consumer created [id:%s]", consumer.id); + }); - await router.createWebRtcTransport({ - listenIps: [{ ip: "127.0.0.1" }], - enableUdp: true, - enableTcp: true, - preferUdp: true + transport.observer.on("newdataproducer", (dataProducer) => { + console.log("new data producer created [id:%s]", dataProducer.id); + }); + + transport.on("trace", (trace) => { + console.log(trace); + }); + + this.mediasoupTransports.push(transport); }); }); diff --git a/webrtc/src/opcodes/Connect.ts b/webrtc/src/opcodes/Connect.ts index 5db11638..b312d6f2 100644 --- a/webrtc/src/opcodes/Connect.ts +++ b/webrtc/src/opcodes/Connect.ts @@ -3,4 +3,8 @@ import { Payload } from "./index"; import { Server } from "../Server" export async function onConnect(this: Server, socket: WebSocket, data: Payload) { + socket.send(JSON.stringify({ + op: 15, + d: { any: 100 } + })) } \ No newline at end of file diff --git a/webrtc/src/opcodes/Identify.ts b/webrtc/src/opcodes/Identify.ts index 6043a460..6bbed04c 100644 --- a/webrtc/src/opcodes/Identify.ts +++ b/webrtc/src/opcodes/Identify.ts @@ -1,9 +1,67 @@ import { WebSocket } from "@fosscord/gateway"; -import { Payload } from "./index" +import { Payload } from "./index"; import { VoiceOPCodes } from "@fosscord/util"; -import { Server } from "../Server" +import { Server } from "../Server"; +import * as mediasoup from "mediasoup"; +import { RtpCodecCapability } from "mediasoup/node/lib/RtpParameters"; + +const test = "extmap-allow-mixed\na=ice-ufrag:ilWh\na=ice-pwd:Mx7TDnPKXDnTgYWC+qMaqspQ\na=ice-options:trickle\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\na=rtpmap:111 opus/48000/2\na=extmap:14 urn:ietf:params:rtp-hdrext:toffset\na=extmap:13 urn:3gpp:video-orientation\na=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\na=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space\na=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\na=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\na=rtpmap:96 VP8/90000\na=rtpmap:97 rtx/90000"; export async function onIdentify(this: Server, socket: WebSocket, data: Payload) { + var transport = await this.mediasoupRouters[0].createWebRtcTransport({ + listenIps: [{ ip: "127.0.0.1" }], + enableUdp: true, + enableTcp: true, + preferUdp: true, + }); + + const rtpCapabilities = this.mediasoupRouters[0].rtpCapabilities; + const codecs = rtpCapabilities.codecs as RtpCodecCapability[]; + + var producer = await transport.produce( + { + kind: "audio", + rtpParameters: + { + mid: "1", + codecs: codecs.filter(x => x.kind === "audio").map((x: RtpCodecCapability) => { + return { + mimeType: x.mimeType, + kind: x.kind, + clockRate: x.clockRate, + channels: x.channels, + payloadType: x.preferredPayloadType as number + }; + }), + headerExtensions: test.split("\na=").map((x, i) => ({ + id: i + 1, + uri: x, + })) + } + }); + + const consumer = await transport.consume( + { + producerId: producer.id, + rtpCapabilities: + { + codecs: codecs.filter(x => x.kind === "audio").map((x: RtpCodecCapability) => { + return { + mimeType: x.mimeType, + kind: x.kind, + clockRate: x.clockRate, + channels: x.channels, + payloadType: x.preferredPayloadType as number + }; + }), + headerExtensions: test.split("\na=").map((x, i) => ({ + kind: "audio", + preferredId: i + 1, + uri: x, + })) + } + }); + socket.send(JSON.stringify({ op: VoiceOPCodes.READY, d: { @@ -11,11 +69,11 @@ export async function onIdentify(this: Server, socket: WebSocket, data: Payload) ip: "127.0.0.1", //@ts-ignore - port: this.mediasoupTransports[0].iceCandidates.port, + port: transport.iceCandidates[0].port, modes: [ "xsalsa20_poly1305", - "xsalsa20_poly1305_suffix", - "xsalsa20_poly1305_lite", + // "xsalsa20_poly1305_suffix", + // "xsalsa20_poly1305_lite", ], heartbeat_interval: 1, }, diff --git a/webrtc/src/opcodes/SelectProtocol.ts b/webrtc/src/opcodes/SelectProtocol.ts index fcc45855..24e8ef5f 100644 --- a/webrtc/src/opcodes/SelectProtocol.ts +++ b/webrtc/src/opcodes/SelectProtocol.ts @@ -1,17 +1,79 @@ import { WebSocket } from "@fosscord/gateway"; import { Payload } from "./index"; import { VoiceOPCodes } from "@fosscord/util"; -import { Server } from "../Server" +import { Server } from "../Server"; + +/* + { + op: 1, + d: { + protocol: "webrtc", + data: " + a=extmap-allow-mixed + a=ice-ufrag:ilWh + a=ice-pwd:Mx7TDnPKXDnTgYWC+qMaqspQ + a=ice-options:trickle + a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level + a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time + a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 + a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid + a=rtpmap:111 opus/48000/2 + a=extmap:14 urn:ietf:params:rtp-hdrext:toffset + a=extmap:13 urn:3gpp:video-orientation + a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay + a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type + a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing + a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space + a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id + a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id + a=rtpmap:96 VP8/90000 + a=rtpmap:97 rtx/90000 + ", + sdp: "same data as in d.data? also not documented by discord", + codecs: [ + { + name: "opus", + type: "audio", + priority: 1000, + payload_type: 111, + rtx_payload_type: null, + }, + { + name: "H264", + type: "video", + priority: 1000, + payload_type: 102, + rtx_payload_type: 121, + }, + { + name: "VP8", + type: "video", + priority: 2000, + payload_type: 96, + rtx_payload_type: 97, + }, + { + name: "VP9", + type: "video", + priority: 3000, + payload_type: 98, + rtx_payload_type: 99, + }, + ], + rtc_connection_id: "b3c8628a-edb5-49ae-b860-ab0d2842b104", + }, + } +*/ export async function onSelectProtocol(this: Server, socket: WebSocket, data: Payload) { socket.send(JSON.stringify({ op: VoiceOPCodes.SESSION_DESCRIPTION, d: { - video_codec: "H264", + video_codec: data.d.codecs.find((x: any) => x.type === "video").name, secret_key: new Array(32).fill(null).map(x => Math.random() * 256), - mode: "aead_aes256_gcm_rtpsize", + mode: "xsalsa20_poly1305", media_session_id: this.mediasoupTransports[0].id, - audio_codec: "opus", + audio_codec: data.d.codecs.find((x: any) => x.type === "audio").name, } })); } \ No newline at end of file diff --git a/webrtc/src/start.ts b/webrtc/src/start.ts index 5614982d..299bfce8 100644 --- a/webrtc/src/start.ts +++ b/webrtc/src/start.ts @@ -1,4 +1,10 @@ +import { config } from "dotenv"; +config(); + import { Server } from "./Server"; +//testing +process.env.DATABASE = "../bundle/database.db"; + const server = new Server(); server.listen(); \ No newline at end of file -- cgit 1.5.1 From b529a372644fab22b34007353cc69caf3de5331c Mon Sep 17 00:00:00 2001 From: Madeline <46743919+MaddyUnderStars@users.noreply.github.com> Date: Mon, 7 Mar 2022 19:15:33 +1100 Subject: augh --- bundle/package.json | 2 +- util/src/util/Constants.ts | 5 +-- webrtc/src/Server.ts | 12 +++++-- webrtc/src/opcodes/Heartbeat.ts | 4 +-- webrtc/src/opcodes/Identify.ts | 37 ++++++++++++++++---- webrtc/src/opcodes/SelectProtocol.ts | 68 ++++++++++++++++++++++++++++-------- webrtc/src/opcodes/Version.ts | 14 ++++++++ webrtc/src/opcodes/index.ts | 3 ++ webrtc/src/start.ts | 6 ++-- webrtc/src/util/Heartbeat.ts | 21 ++++++----- 10 files changed, 133 insertions(+), 39 deletions(-) create mode 100644 webrtc/src/opcodes/Version.ts (limited to 'webrtc/src/start.ts') diff --git a/bundle/package.json b/bundle/package.json index 0b00b325..aedd963b 100644 --- a/bundle/package.json +++ b/bundle/package.json @@ -112,4 +112,4 @@ "typescript-json-schema": "^0.50.1", "ws": "^7.4.2" } -} +} \ No newline at end of file diff --git a/util/src/util/Constants.ts b/util/src/util/Constants.ts index d5315767..42a2c274 100644 --- a/util/src/util/Constants.ts +++ b/util/src/util/Constants.ts @@ -77,8 +77,9 @@ export const VoiceOPCodes = { RESUME: 7, HELLO: 8, RESUMED: 9, - CLIENT_CONNECT: 12, - CLIENT_DISCONNECT: 13, + CLIENT_CONNECT: 12, // incorrect, op 12 is probably used for video + CLIENT_DISCONNECT: 13, // incorrect + VERSION: 16, //not documented }; export const Events = { diff --git a/webrtc/src/Server.ts b/webrtc/src/Server.ts index 0145a221..1d18d6d1 100644 --- a/webrtc/src/Server.ts +++ b/webrtc/src/Server.ts @@ -6,6 +6,8 @@ import { setHeartbeat } from "./util"; import * as mediasoup from "mediasoup"; import { types as MediasoupTypes } from "mediasoup"; +import Net from "net"; + var port = Number(process.env.PORT); if (isNaN(port)) port = 3004; @@ -13,7 +15,7 @@ export class Server { public ws: WebSocketServer; public mediasoupWorkers: MediasoupTypes.Worker[] = []; public mediasoupRouters: MediasoupTypes.Router[] = []; - public mediasoupTransports: MediasoupTypes.Transport[] = []; + public mediasoupTransports: MediasoupTypes.WebRtcTransport[] = []; constructor() { this.ws = new WebSocketServer({ @@ -26,7 +28,7 @@ export class Server { socket.on("message", async (message: string) => { const payload: Payload = JSON.parse(message); - console.log(payload); + // console.log(payload); if (OPCodeHandlers[payload.op]) try { @@ -68,9 +70,13 @@ export class Server { this.mediasoupRouters.push(router); - router.observer.on("newtransport", async (transport: MediasoupTypes.Transport) => { + router.observer.on("newtransport", async (transport: MediasoupTypes.WebRtcTransport) => { console.log("new transport created [id:%s]", transport.id); + transport.observer.on("sctpstatechange", (state) => { + console.log(state) + }); + await transport.enableTraceEvent(); transport.observer.on("newproducer", (producer: MediasoupTypes.Producer) => { diff --git a/webrtc/src/opcodes/Heartbeat.ts b/webrtc/src/opcodes/Heartbeat.ts index 06d6bcb1..47f33f76 100644 --- a/webrtc/src/opcodes/Heartbeat.ts +++ b/webrtc/src/opcodes/Heartbeat.ts @@ -1,8 +1,8 @@ import { WebSocket } from "@fosscord/gateway"; import { Payload } from "./index"; -import { setHeartbeat } from "./../util"; +import { setHeartbeat } from "../util"; import { Server } from "../Server" export async function onHeartbeat(this: Server, socket: WebSocket, data: Payload) { - await setHeartbeat(socket); + await setHeartbeat(socket, data.d); } \ No newline at end of file diff --git a/webrtc/src/opcodes/Identify.ts b/webrtc/src/opcodes/Identify.ts index e965e3de..d7da5c7c 100644 --- a/webrtc/src/opcodes/Identify.ts +++ b/webrtc/src/opcodes/Identify.ts @@ -28,12 +28,12 @@ export async function onIdentify(this: Server, socket: WebSocket, data: Identify } ); const user = session.user; - const guild = await Guild.findOneOrFail({ id: data.d.server_id }); + const guild = await Guild.findOneOrFail({ id: data.d.server_id }, { relations: ["members"] }); if (!guild.members.find(x => x.id === user.id)) return socket.close(CLOSECODES.Invalid_intent); - var transport = await this.mediasoupRouters[0].createWebRtcTransport({ + var transport = this.mediasoupTransports[0] || await this.mediasoupRouters[0].createWebRtcTransport({ listenIps: [{ ip: "0.0.0.0", announcedIp: "127.0.0.1" }], enableUdp: true, enableTcp: true, @@ -66,13 +66,39 @@ export async function onIdentify(this: Server, socket: WebSocket, data: Identify } */ + + + /* + { + "streams": [ + { "type": "video", "ssrc": 129861, "rtx_ssrc": 129862, "rid": "100", "quality": 100, "active": false } + ], + "ssrc": 129860, + "port": 50003, + "modes": [ + "aead_aes256_gcm_rtpsize", + "aead_aes256_gcm", + "xsalsa20_poly1305_lite_rtpsize", + "xsalsa20_poly1305_lite", + "xsalsa20_poly1305_suffix", + "xsalsa20_poly1305" + ], + "ip": "109.200.213.251", + "experiments": [ + "bwe_conservative_link_estimate", + "bwe_remote_locus_client", + "fixed_keyframe_interval" + ]; + }; + */ + socket.send(JSON.stringify({ op: VoiceOPCodes.READY, d: { - streams: [...data.d.streams.map(x => ({ ...x, rtx_ssrc: 1311886, ssrc: 1311885, active: false, }))], - ssrc: 1, + streams: [...data.d.streams.map(x => ({ ...x, rtx_ssrc: Math.floor(Math.random() * 10000), ssrc: Math.floor(Math.random() * 10000), active: false, }))], + ssrc: Math.floor(Math.random() * 10000), ip: transport.iceCandidates[0].ip, - port: transport.iceCandidates[0].port, + port: "50001", modes: [ "aead_aes256_gcm_rtpsize", "aead_aes256_gcm", @@ -81,7 +107,6 @@ export async function onIdentify(this: Server, socket: WebSocket, data: Identify "xsalsa20_poly1305_suffix", "xsalsa20_poly1305" ], - heartbeat_interval: 1, experiments: [], }, })); diff --git a/webrtc/src/opcodes/SelectProtocol.ts b/webrtc/src/opcodes/SelectProtocol.ts index 36527a8b..a957e14f 100644 --- a/webrtc/src/opcodes/SelectProtocol.ts +++ b/webrtc/src/opcodes/SelectProtocol.ts @@ -87,42 +87,82 @@ export async function onSelectProtocol(this: Server, socket: WebSocket, data: Pa })), */ + const videoCodec = this.mediasoupRouters[0].rtpCapabilities.codecs!.find((x: any) => x.kind === "video")?.mimeType + const audioCodec = this.mediasoupRouters[0].rtpCapabilities.codecs!.find((x: any) => x.kind === "audio") + if (!test_hasMadeProducer) { const producer = await transport.produce({ kind: "audio", rtpParameters: { mid: "audio", codecs: [{ - clockRate: 48000, - payloadType: 111, - mimeType: "audio/opus", - channels: 2, + clockRate: audioCodec!.clockRate, + payloadType: audioCodec!.preferredPayloadType as number, + mimeType: audioCodec!.mimeType, + channels: audioCodec?.channels, }], headerExtensions: res.ext?.map(x => ({ id: x.value, uri: x.uri, - })) + })), }, paused: false, }); - + const consumer = await transport.consume({ producerId: producer.id, - paused: false, + paused: true, rtpCapabilities, - }) - + }); + test_hasMadeProducer = true; } + /* server sends sdp: + + m=audio 50021 ICE/SDP //same port as sent in READY + a=fingerprint:sha-256 4A:79:94:16:44:3F:BD:05:41:5A:C7:20:F3:12:54:70:00:73:5D:33:00:2D:2C:80:9B:39:E1:9F:2D:A7:49:87 + c=IN IP4 109.200.213.132 //same IP as sent in READY + a=rtcp:50021 //same port? + a=ice-ufrag:rTmX + a=ice-pwd:M+ncqWK6SEdHhirOjG2VFA + a=fingerprint:sha-256 4A:79:94:16:44:3F:BD:05:41:5A:C7:20:F3:12:54:70:00:73:5D:33:00:2D:2C:80:9B:39:E1:9F:2D:A7:49:87 + a=candidate:1 1 UDP 4261412862 109.200.213.132 50021 typ host //same IP and PORT + + */ + + + var test = { + "video_codec": "H264", + "sdp": ` + m=audio 50011 ICE/SDP\n + a=fingerprint:sha-256 4A:79:94:16:44:3F:BD:05:41:5A:C7:20:F3:12:54:70:00:73:5D:33:00:2D:2C:80:9B:39:E1:9F:2D:A7:49:87\n + c=IN IP4 109.200.214.156\n + a=rtcp:50011\n + a=ice-ufrag:d0aZ\n + a=ice-pwd:51ubWYu7GSkQRqlH/apTSZ\n + a=fingerprint:sha-256 4A:79:94:16:44:3F:BD:05:41:5A:C7:20:F3:12:54:70:00:73:5D:33:00:2D:2C:80:9B:39:E1:9F:2D:A7:49:87\n + a=candidate:1 1 UDP 4261412862 109.200.214.156 50011 typ host\n`, + "media_session_id": "9e18c981687f2de5399edd5cb3f3babf", + "audio_codec": "opus" + }; + + socket.send(JSON.stringify({ op: VoiceOPCodes.SESSION_DESCRIPTION, d: { - video_codec: data.d.codecs.find((x: any) => x.type === "video").name, - secret_key: new Array(32).fill(null).map(x => Math.random() * 256), - mode: "xsalsa20_poly1305", - media_session_id: this.mediasoupTransports[0].id, - audio_codec: data.d.codecs.find((x: any) => x.type === "audio").name, + video_codec: videoCodec?.substring(6) || undefined, + // mode: "xsalsa20_poly1305", + media_session_id: transport.id, + audio_codec: audioCodec?.mimeType.substring(6), + sdp: `m=audio ${transport.iceCandidates[0].port} ICE/SDP\n` + + `a=fingerprint:sha-256 ${transport.dtlsParameters.fingerprints.find(x => x.algorithm === "sha-256")?.value}\n` + + `c=IN IPV4 ${transport.iceCandidates[0].ip}\n` + + `a=rtcp:${transport.iceCandidates[0].port}\n` + + `a=ice-ufrag:${transport.iceParameters.usernameFragment}\n` + + `a=ice-pwd:${transport.iceParameters.password}\n` + + `a=fingerprint:sha-1 ${transport.dtlsParameters.fingerprints[0].value}\n` + + `a=candidate:1 1 ${transport.iceCandidates[0].protocol} ${transport.iceCandidates[0].priority} ${transport.iceCandidates[0].ip} ${transport.iceCandidates[0].port} typ ${transport.iceCandidates[0].type}` } })); } \ No newline at end of file diff --git a/webrtc/src/opcodes/Version.ts b/webrtc/src/opcodes/Version.ts new file mode 100644 index 00000000..0ea6eb4d --- /dev/null +++ b/webrtc/src/opcodes/Version.ts @@ -0,0 +1,14 @@ +import { WebSocket } from "@fosscord/gateway"; +import { Payload } from "./index"; +import { setHeartbeat } from "../util"; +import { Server } from "../Server" + +export async function onVersion(this: Server, socket: WebSocket, data: Payload) { + socket.send(JSON.stringify({ + op: 16, + d: { + voice: "0.8.31", //version numbers? + rtc_worker: "0.3.18", + } + })) +} \ No newline at end of file diff --git a/webrtc/src/opcodes/index.ts b/webrtc/src/opcodes/index.ts index 9b1eb270..d0f40bc2 100644 --- a/webrtc/src/opcodes/index.ts +++ b/webrtc/src/opcodes/index.ts @@ -15,6 +15,8 @@ import { onSpeaking } from "./Speaking"; import { onResume } from "./Resume"; import { onConnect } from "./Connect"; +import { onVersion } from "./Version"; + export type OPCodeHandler = (this: WebSocket, data: Payload) => any; export default { @@ -34,4 +36,5 @@ export default { //op 13? //op 15? //op 16? empty data on client send but server sends {"voice":"0.8.24+bugfix.voice.streams.opt.branch-ffcefaff7","rtc_worker":"0.3.14-crypto-collision-copy"} + [VoiceOPCodes.VERSION]: onVersion, }; \ No newline at end of file diff --git a/webrtc/src/start.ts b/webrtc/src/start.ts index 299bfce8..98f06ad5 100644 --- a/webrtc/src/start.ts +++ b/webrtc/src/start.ts @@ -1,10 +1,10 @@ +//testing +process.env.DATABASE = "../bundle/database.db"; + import { config } from "dotenv"; config(); import { Server } from "./Server"; -//testing -process.env.DATABASE = "../bundle/database.db"; - const server = new Server(); server.listen(); \ No newline at end of file diff --git a/webrtc/src/util/Heartbeat.ts b/webrtc/src/util/Heartbeat.ts index 7b5ed9cd..8c5e3a7a 100644 --- a/webrtc/src/util/Heartbeat.ts +++ b/webrtc/src/util/Heartbeat.ts @@ -1,18 +1,23 @@ import { WebSocket, CLOSECODES } from "@fosscord/gateway"; import { VoiceOPCodes } from "@fosscord/util"; -export async function setHeartbeat(socket: WebSocket) { +export async function setHeartbeat(socket: WebSocket, nonce?: Number) { if (socket.heartbeatTimeout) clearTimeout(socket.heartbeatTimeout); socket.heartbeatTimeout = setTimeout(() => { return socket.close(CLOSECODES.Session_timed_out); }, 1000 * 45); - socket.send(JSON.stringify({ - op: VoiceOPCodes.HEARTBEAT_ACK, - d: { - v: 6, - heartbeat_interval: 13750, - } - })); + if (!nonce) { + socket.send(JSON.stringify({ + op: VoiceOPCodes.HELLO, + d: { + v: 5, + heartbeat_interval: 13750, + } + })); + } + else { + socket.send(JSON.stringify({ op: VoiceOPCodes.HEARTBEAT_ACK, d: nonce })); + } } \ No newline at end of file -- cgit 1.5.1