diff --git a/webrtc/src/Server.ts b/webrtc/src/Server.ts
index 6591691c..dcbf216a 100644
--- a/webrtc/src/Server.ts
+++ b/webrtc/src/Server.ts
@@ -1,46 +1,102 @@
import { Server as WebSocketServer } from "ws";
-import { Config, db } from "@fosscord/util";
-import mediasoup from "mediasoup";
+import { WebSocket, Payload, CLOSECODES } from "@fosscord/gateway";
+import { Config, initDatabase } from "@fosscord/util";
+import OPCodeHandlers from "./opcodes";
+import { setHeartbeat } from "./util";
+import * as mediasoup from "mediasoup";
+import { types as MediasoupTypes } from "mediasoup";
var port = Number(process.env.PORT);
if (isNaN(port)) port = 3004;
export class Server {
public ws: WebSocketServer;
- public turn: any;
+ public mediasoupWorkers: MediasoupTypes.Worker[] = [];
+ public mediasoupRouters: MediasoupTypes.Router[] = [];
+ public mediasoupTransports: MediasoupTypes.Transport[] = [];
constructor() {
this.ws = new WebSocketServer({
port,
maxPayload: 4096,
});
- this.ws.on("connection", (socket) => {
- socket.on("message", (message) => {
- socket.emit(
- JSON.stringify({
- op: 2,
- d: {
- ssrc: 1,
- ip: "127.0.0.1",
- port: 3004,
- modes: [
- "xsalsa20_poly1305",
- "xsalsa20_poly1305_suffix",
- "xsalsa20_poly1305_lite",
- ],
- heartbeat_interval: 1,
- },
- })
- );
+ this.ws.on("connection", async (socket: WebSocket) => {
+ await setHeartbeat(socket);
+
+ socket.on("message", async (message: string) => {
+ const payload: Payload = JSON.parse(message);
+
+ if (OPCodeHandlers[payload.op])
+ await OPCodeHandlers[payload.op].call(this, socket, payload);
+ else {
+ console.error(`Unimplemented`, payload);
+ socket.close(CLOSECODES.Unknown_opcode);
+ }
});
});
}
async listen(): Promise<void> {
// @ts-ignore
- await (db as Promise<Connection>);
+ await initDatabase();
await Config.init();
+ await this.createWorkers();
console.log("[DB] connected");
console.log(`[WebRTC] online on 0.0.0.0:${port}`);
}
+
+ async createWorkers(): Promise<void> {
+ const numWorkers = 1;
+ for (let i = 0; i < numWorkers; i++) {
+ const worker = await mediasoup.createWorker({ logLevel: "debug" });
+ if (!worker) return;
+
+ worker.on("died", () => {
+ console.error("mediasoup worker died");
+ });
+
+ worker.observer.on("newrouter", async (router: MediasoupTypes.Router) => {
+ console.log("new router created [id:%s]", router.id);
+
+ this.mediasoupRouters.push(router);
+
+ router.observer.on("newtransport", async (transport: MediasoupTypes.Transport) => {
+ console.log("new transport created [id:%s]", transport.id);
+
+ 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);
+ });
+
+ 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);
+ });
+ });
+
+ await worker.createRouter({
+ mediaCodecs: [
+ {
+ kind: "audio",
+ mimeType: "audio/opus",
+ clockRate: 48000,
+ channels: 2
+ },
+ ]
+ });
+
+ this.mediasoupWorkers.push(worker);
+ }
+ }
}
diff --git a/webrtc/src/opcodes/Connect.ts b/webrtc/src/opcodes/Connect.ts
new file mode 100644
index 00000000..b312d6f2
--- /dev/null
+++ b/webrtc/src/opcodes/Connect.ts
@@ -0,0 +1,10 @@
+import { WebSocket } from "@fosscord/gateway";
+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/Heartbeat.ts b/webrtc/src/opcodes/Heartbeat.ts
new file mode 100644
index 00000000..06d6bcb1
--- /dev/null
+++ b/webrtc/src/opcodes/Heartbeat.ts
@@ -0,0 +1,8 @@
+import { WebSocket } from "@fosscord/gateway";
+import { Payload } from "./index";
+import { setHeartbeat } from "./../util";
+import { Server } from "../Server"
+
+export async function onHeartbeat(this: Server, socket: WebSocket, data: Payload) {
+ await setHeartbeat(socket);
+}
\ No newline at end of file
diff --git a/webrtc/src/opcodes/Identify.ts b/webrtc/src/opcodes/Identify.ts
new file mode 100644
index 00000000..82f327be
--- /dev/null
+++ b/webrtc/src/opcodes/Identify.ts
@@ -0,0 +1,57 @@
+import { WebSocket } from "@fosscord/gateway";
+import { Payload } from "./index";
+import { VoiceOPCodes } from "@fosscord/util";
+import { Server } from "../Server";
+
+export async function onIdentify(this: Server, socket: WebSocket, data: Payload) {
+ var transport = await this.mediasoupRouters[0].createWebRtcTransport({
+ listenIps: [{ ip: "0.0.0.0", announcedIp: "127.0.0.1" }],
+ enableUdp: true,
+ enableTcp: true,
+ preferUdp: true,
+ });
+
+ /*
+ //discord proper sends:
+ {
+ "streams": [
+ { "type": "video", "ssrc": 1311885, "rtx_ssrc": 1311886, "rid": "50", "quality": 50, "active": false },
+ { "type": "video", "ssrc": 1311887, "rtx_ssrc": 1311888, "rid": "100", "quality": 100, "active": false }
+ ],
+ "ssrc": 1311884,
+ "port": 50008,
+ "modes": [
+ "aead_aes256_gcm_rtpsize",
+ "aead_aes256_gcm",
+ "xsalsa20_poly1305_lite_rtpsize",
+ "xsalsa20_poly1305_lite",
+ "xsalsa20_poly1305_suffix",
+ "xsalsa20_poly1305"
+ ],
+ "ip": "109.200.214.158",
+ "experiments": [
+ "bwe_conservative_link_estimate",
+ "bwe_remote_locus_client",
+ "fixed_keyframe_interval"
+ ]
+ }
+ */
+
+ socket.send(JSON.stringify({
+ op: VoiceOPCodes.READY,
+ d: {
+ streams: [],
+ ssrc: 1,
+ ip: transport.iceCandidates[0].ip,
+ port: transport.iceCandidates[0].port,
+ modes: [
+ "aead_aes256_gcm_rtpsize",
+ // "xsalsa20_poly1305",
+ // "xsalsa20_poly1305_suffix",
+ // "xsalsa20_poly1305_lite",
+ ],
+ heartbeat_interval: 1,
+ experiments: [],
+ },
+ }));
+}
\ No newline at end of file
diff --git a/webrtc/src/opcodes/Resume.ts b/webrtc/src/opcodes/Resume.ts
new file mode 100644
index 00000000..dcd4f4cd
--- /dev/null
+++ b/webrtc/src/opcodes/Resume.ts
@@ -0,0 +1,6 @@
+import { WebSocket } from "@fosscord/gateway";
+import { Payload } from "./index";
+import { Server } from "../Server"
+
+export async function onResume(this: Server, socket: WebSocket, data: Payload) {
+}
\ No newline at end of file
diff --git a/webrtc/src/opcodes/SelectProtocol.ts b/webrtc/src/opcodes/SelectProtocol.ts
new file mode 100644
index 00000000..36527a8b
--- /dev/null
+++ b/webrtc/src/opcodes/SelectProtocol.ts
@@ -0,0 +1,128 @@
+import { WebSocket } from "@fosscord/gateway";
+import { Payload } from "./index";
+import { VoiceOPCodes } from "@fosscord/util";
+import { Server } from "../Server";
+import * as mediasoup from "mediasoup";
+import { RtpCodecCapability } from "mediasoup/node/lib/RtpParameters";
+import * as sdpTransform from 'sdp-transform';
+
+/*
+ {
+ 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",
+ },
+ }
+*/
+
+var test_hasMadeProducer = false;
+
+export async function onSelectProtocol(this: Server, socket: WebSocket, data: Payload) {
+ const rtpCapabilities = this.mediasoupRouters[0].rtpCapabilities;
+ const codecs = rtpCapabilities.codecs as RtpCodecCapability[];
+
+ const transport = this.mediasoupTransports[0]; //whatever
+
+ const res = sdpTransform.parse(data.d.sdp);
+
+ /*
+ res.media.map(x => x.rtp).flat(1).map(x => ({
+ codec: x.codec,
+ payloadType: x.payload,
+ clockRate: x.rate as number,
+ mimeType: `audio/${x.codec}`,
+ })),
+ */
+
+ if (!test_hasMadeProducer) {
+ const producer = await transport.produce({
+ kind: "audio",
+ rtpParameters: {
+ mid: "audio",
+ codecs: [{
+ clockRate: 48000,
+ payloadType: 111,
+ mimeType: "audio/opus",
+ channels: 2,
+ }],
+ headerExtensions: res.ext?.map(x => ({
+ id: x.value,
+ uri: x.uri,
+ }))
+ },
+ paused: false,
+ });
+
+ const consumer = await transport.consume({
+ producerId: producer.id,
+ paused: false,
+ rtpCapabilities,
+ })
+
+ test_hasMadeProducer = true;
+ }
+
+ 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,
+ }
+ }));
+}
\ No newline at end of file
diff --git a/webrtc/src/opcodes/Speaking.ts b/webrtc/src/opcodes/Speaking.ts
new file mode 100644
index 00000000..861a7c3d
--- /dev/null
+++ b/webrtc/src/opcodes/Speaking.ts
@@ -0,0 +1,7 @@
+import { WebSocket } from "@fosscord/gateway";
+import { Payload } from "./index"
+import { VoiceOPCodes } from "@fosscord/util";
+import { Server } from "../Server"
+
+export async function onSpeaking(this: Server, socket: WebSocket, data: Payload) {
+}
\ No newline at end of file
diff --git a/webrtc/src/opcodes/index.ts b/webrtc/src/opcodes/index.ts
new file mode 100644
index 00000000..36d30e7d
--- /dev/null
+++ b/webrtc/src/opcodes/index.ts
@@ -0,0 +1,37 @@
+import { WebSocket } from "@fosscord/gateway";
+import { VoiceOPCodes } from "@fosscord/util";
+
+export interface Payload {
+ op: number;
+ d?: any;
+ s?: number;
+ t?: string;
+}
+
+import { onIdentify } from "./Identify";
+import { onSelectProtocol } from "./SelectProtocol";
+import { onHeartbeat } from "./Heartbeat";
+import { onSpeaking } from "./Speaking";
+import { onResume } from "./Resume";
+import { onConnect } from "./Connect";
+
+export type OPCodeHandler = (this: WebSocket, data: Payload) => any;
+
+export default {
+ [VoiceOPCodes.IDENTIFY]: onIdentify, //op 0
+ [VoiceOPCodes.SELECT_PROTOCOL]: onSelectProtocol, //op 1
+ //op 2 voice_ready
+ [VoiceOPCodes.HEARTBEAT]: onHeartbeat, //op 3
+ //op 4 session_description
+ [VoiceOPCodes.SPEAKING]: onSpeaking, //op 5
+ //op 6 heartbeat_ack
+ [VoiceOPCodes.RESUME]: onResume, //op 7
+ //op 8 hello
+ //op 9 resumed
+ //op 10?
+ //op 11?
+ [VoiceOPCodes.CLIENT_CONNECT]: onConnect, //op 12
+ //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"}
+};
\ No newline at end of file
diff --git a/webrtc/src/start.ts b/webrtc/src/start.ts
index 68867a2c..299bfce8 100644
--- a/webrtc/src/start.ts
+++ b/webrtc/src/start.ts
@@ -1,3 +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
diff --git a/webrtc/src/util/Heartbeat.ts b/webrtc/src/util/Heartbeat.ts
new file mode 100644
index 00000000..7b5ed9cd
--- /dev/null
+++ b/webrtc/src/util/Heartbeat.ts
@@ -0,0 +1,18 @@
+import { WebSocket, CLOSECODES } from "@fosscord/gateway";
+import { VoiceOPCodes } from "@fosscord/util";
+
+export async function setHeartbeat(socket: WebSocket) {
+ 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,
+ }
+ }));
+}
\ No newline at end of file
diff --git a/webrtc/src/util/index.ts b/webrtc/src/util/index.ts
new file mode 100644
index 00000000..e8557452
--- /dev/null
+++ b/webrtc/src/util/index.ts
@@ -0,0 +1 @@
+export * from "./Heartbeat"
\ No newline at end of file
|
