diff --git a/webrtc/src/Server.ts b/webrtc/src/Server.ts
index 6591691c..42b82c6a 100644
--- a/webrtc/src/Server.ts
+++ b/webrtc/src/Server.ts
@@ -1,46 +1,125 @@
import { Server as WebSocketServer } from "ws";
-import { Config, db } from "@fosscord/util";
-import mediasoup from "mediasoup";
+import { WebSocket, CLOSECODES } from "@fosscord/gateway";
+import { Config, initDatabase } from "@fosscord/util";
+import OPCodeHandlers, { Payload } from "./opcodes";
+import { setHeartbeat } from "./util";
+import * as mediasoup from "mediasoup";
+import { types as MediasoupTypes } from "mediasoup";
+
+import udp from "dgram";
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.WebRtcTransport[] = [];
+ public mediasoupProducers: MediasoupTypes.Producer[] = [];
+ public mediasoupConsumers: MediasoupTypes.Consumer[] = [];
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])
+ try {
+ await OPCodeHandlers[payload.op].call(this, socket, payload);
+ }
+ catch (e) {
+ console.error(e);
+ socket.close(CLOSECODES.Unknown_error);
+ }
+ 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.WebRtcTransport) => {
+ console.log("new transport created [id:%s]", transport.id);
+
+ await transport.enableTraceEvent();
+
+ transport.on("connect", () => {
+ console.log("transport connect")
+ })
+
+ transport.observer.on("newproducer", (producer: MediasoupTypes.Producer) => {
+ console.log("new producer created [id:%s]", producer.id);
+
+ this.mediasoupProducers.push(producer);
+ });
+
+ transport.observer.on("newconsumer", (consumer: MediasoupTypes.Consumer) => {
+ console.log("new consumer created [id:%s]", consumer.id);
+
+ this.mediasoupConsumers.push(consumer);
+
+ consumer.on("rtp", (rtpPacket) => {
+ console.log(rtpPacket);
+ });
+ });
+
+ 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..1f874a44
--- /dev/null
+++ b/webrtc/src/opcodes/Connect.ts
@@ -0,0 +1,40 @@
+import { WebSocket } from "@fosscord/gateway";
+import { Payload } from "./index";
+import { Server } from "../Server"
+
+/*
+Sent by client:
+
+{
+ "op": 12,
+ "d": {
+ "audio_ssrc": 0,
+ "video_ssrc": 0,
+ "rtx_ssrc": 0,
+ "streams": [
+ {
+ "type": "video",
+ "rid": "100",
+ "ssrc": 0,
+ "active": false,
+ "quality": 100,
+ "rtx_ssrc": 0,
+ "max_bitrate": 2500000,
+ "max_framerate": 20,
+ "max_resolution": {
+ "type": "fixed",
+ "width": 1280,
+ "height": 720
+ }
+ }
+ ]
+ }
+}
+*/
+
+export async function onConnect(this: Server, socket: WebSocket, data: Payload) {
+ socket.send(JSON.stringify({ //what is op 15?
+ 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..47f33f76
--- /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, data.d);
+}
\ 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..9baa16e3
--- /dev/null
+++ b/webrtc/src/opcodes/Identify.ts
@@ -0,0 +1,66 @@
+import { WebSocket, CLOSECODES } from "@fosscord/gateway";
+import { Payload } from "./index";
+import { VoiceOPCodes, Session, User, Guild } from "@fosscord/util";
+import { Server } from "../Server";
+
+export interface IdentifyPayload extends Payload {
+ d: {
+ server_id: string, //guild id
+ session_id: string, //gateway session
+ streams: Array<{
+ type: string,
+ rid: string, //number
+ quality: number,
+ }>,
+ token: string, //voice_states token
+ user_id: string,
+ video: boolean,
+ };
+}
+
+export async function onIdentify(this: Server, socket: WebSocket, data: IdentifyPayload) {
+
+ const session = await Session.findOneOrFail(
+ { session_id: data.d.session_id, },
+ {
+ where: { user_id: data.d.user_id },
+ relations: ["user"]
+ }
+ );
+ const user = session.user;
+ 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 = this.mediasoupTransports[0] || await this.mediasoupRouters[0].createWebRtcTransport({
+ listenIps: [{ ip: "10.22.64.69" }],
+ enableUdp: true,
+ enableTcp: true,
+ preferUdp: true,
+ enableSctp: true,
+ });
+
+ socket.send(JSON.stringify({
+ op: VoiceOPCodes.READY,
+ d: {
+ 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,
+ modes: [
+ "aead_aes256_gcm_rtpsize",
+ "aead_aes256_gcm",
+ "xsalsa20_poly1305_lite_rtpsize",
+ "xsalsa20_poly1305_lite",
+ "xsalsa20_poly1305_suffix",
+ "xsalsa20_poly1305"
+ ],
+ experiments: [
+ "bwe_conservative_link_estimate",
+ "bwe_remote_locus_client",
+ "fixed_keyframe_interval"
+ ]
+ },
+ }));
+}
\ 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..856b550c
--- /dev/null
+++ b/webrtc/src/opcodes/Resume.ts
@@ -0,0 +1,24 @@
+import { CLOSECODES, WebSocket } from "@fosscord/gateway";
+import { Payload } from "./index";
+import { Server } from "../Server"
+import { Guild, Session, VoiceOPCodes } from "@fosscord/util";
+
+export async function onResume(this: Server, socket: WebSocket, data: Payload) {
+ const session = await Session.findOneOrFail(
+ { session_id: data.d.session_id, },
+ {
+ where: { user_id: data.d.user_id },
+ relations: ["user"]
+ }
+ );
+ const user = session.user;
+ 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);
+
+ socket.send(JSON.stringify({
+ op: VoiceOPCodes.RESUMED,
+ d: null,
+ }))
+}
\ 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..dc9d2b88
--- /dev/null
+++ b/webrtc/src/opcodes/SelectProtocol.ts
@@ -0,0 +1,150 @@
+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';
+
+
+/*
+
+ Sent by client:
+{
+ "op": 1,
+ "d": {
+ "protocol": "webrtc",
+ "data": "
+ a=extmap-allow-mixed
+ a=ice-ufrag:vNxb
+ a=ice-pwd:tZvpbVPYEKcnW0gGRPq0OOnh
+ 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
+ ",
+ "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": "3faa0b80-b3e2-4bae-b291-273801fbb7ab"
+ }
+}
+
+Sent by server:
+
+{
+ "op": 4,
+ "d": {
+ "video_codec": "H264",
+ "sdp": "
+ m=audio 50001 ICE/SDP
+ 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 109.200.214.158
+ a=rtcp:50001
+ a=ice-ufrag:CLzn
+ a=ice-pwd:qEmIcNwigd07mu46Ok0XCh
+ 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 109.200.214.158 50001 typ host
+ ",
+ "media_session_id": "807955cb953e98c5b90704cf048e81ec",
+ "audio_codec": "opus"
+ }
+}
+
+*/
+
+
+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);
+
+ const videoCodec = this.mediasoupRouters[0].rtpCapabilities.codecs!.find((x: any) => x.kind === "video");
+ const audioCodec = this.mediasoupRouters[0].rtpCapabilities.codecs!.find((x: any) => x.kind === "audio");
+
+ const producer = this.mediasoupProducers[0] || await transport.produce({
+ kind: "audio",
+ rtpParameters: {
+ mid: "audio",
+ codecs: [{
+ 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,
+ });
+
+ console.log("can consume: " + this.mediasoupRouters[0].canConsume({ producerId: producer.id, rtpCapabilities: rtpCapabilities }));
+
+ const consumer = this.mediasoupConsumers[0] || await transport.consume({
+ producerId: producer.id,
+ paused: false,
+ rtpCapabilities,
+ });
+
+ socket.send(JSON.stringify({
+ op: VoiceOPCodes.SESSION_DESCRIPTION,
+ d: {
+ video_codec: videoCodec?.mimeType?.substring(6) || undefined,
+ mode: "xsalsa20_poly1305_lite",
+ 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/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/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
new file mode 100644
index 00000000..d0f40bc2
--- /dev/null
+++ b/webrtc/src/opcodes/index.ts
@@ -0,0 +1,40 @@
+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";
+
+import { onVersion } from "./Version";
+
+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"}
+ [VoiceOPCodes.VERSION]: onVersion,
+};
\ No newline at end of file
diff --git a/webrtc/src/start.ts b/webrtc/src/start.ts
index 68867a2c..98f06ad5 100644
--- a/webrtc/src/start.ts
+++ b/webrtc/src/start.ts
@@ -1,3 +1,10 @@
+//testing
+process.env.DATABASE = "../bundle/database.db";
+
+import { config } from "dotenv";
+config();
+
import { Server } from "./Server";
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..8c5e3a7a
--- /dev/null
+++ b/webrtc/src/util/Heartbeat.ts
@@ -0,0 +1,23 @@
+import { WebSocket, CLOSECODES } from "@fosscord/gateway";
+import { VoiceOPCodes } from "@fosscord/util";
+
+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);
+
+ 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
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
|
