diff --git a/src/webrtc/Server.ts b/src/webrtc/Server.ts
index 32b795ea..d9a892a3 100644
--- a/src/webrtc/Server.ts
+++ b/src/webrtc/Server.ts
@@ -11,7 +11,15 @@ export class Server {
public server: http.Server;
public production: boolean;
- constructor({ port, server, production }: { port: number; server?: http.Server; production?: boolean }) {
+ constructor({
+ port,
+ server,
+ production,
+ }: {
+ port: number;
+ server?: http.Server;
+ production?: boolean;
+ }) {
this.port = port;
this.production = production || false;
@@ -53,4 +61,4 @@ export class Server {
closeDatabase();
this.server.close();
}
-}
\ No newline at end of file
+}
diff --git a/src/webrtc/events/Close.ts b/src/webrtc/events/Close.ts
index 1c203653..4cf80bb2 100644
--- a/src/webrtc/events/Close.ts
+++ b/src/webrtc/events/Close.ts
@@ -6,4 +6,4 @@ export async function onClose(this: WebSocket, code: number, reason: string) {
if (this.session_id) await Session.delete({ session_id: this.session_id });
this.removeAllListeners();
-}
\ No newline at end of file
+}
diff --git a/src/webrtc/events/Connection.ts b/src/webrtc/events/Connection.ts
index bf228d64..9300b6b2 100644
--- a/src/webrtc/events/Connection.ts
+++ b/src/webrtc/events/Connection.ts
@@ -14,7 +14,11 @@ try {
// TODO: specify rate limit in config
// TODO: check msg max size
-export async function Connection(this: WS.Server, socket: WebSocket, request: IncomingMessage) {
+export async function Connection(
+ this: WS.Server,
+ socket: WebSocket,
+ request: IncomingMessage,
+) {
try {
socket.on("close", onClose.bind(socket));
socket.on("message", onMessage.bind(socket));
@@ -29,7 +33,7 @@ export async function Connection(this: WS.Server, socket: WebSocket, request: In
"open",
"ping",
"pong",
- "unexpected-response"
+ "unexpected-response",
].forEach((x) => {
socket.on(x, (y) => console.log("[WebRTC]", x, y));
});
@@ -39,7 +43,8 @@ export async function Connection(this: WS.Server, socket: WebSocket, request: In
socket.encoding = "json";
socket.version = Number(searchParams.get("v")) || 5;
- if (socket.version < 3) return socket.close(CLOSECODES.Unknown_error, "invalid version");
+ if (socket.version < 3)
+ return socket.close(CLOSECODES.Unknown_error, "invalid version");
setHeartbeat(socket);
@@ -50,11 +55,11 @@ export async function Connection(this: WS.Server, socket: WebSocket, request: In
await Send(socket, {
op: VoiceOPCodes.HELLO,
d: {
- heartbeat_interval: 1000 * 30
- }
+ heartbeat_interval: 1000 * 30,
+ },
});
} catch (error) {
console.error("[WebRTC]", error);
return socket.close(CLOSECODES.Unknown_error);
}
-}
\ No newline at end of file
+}
diff --git a/src/webrtc/events/Message.ts b/src/webrtc/events/Message.ts
index 8f75a815..38676f6c 100644
--- a/src/webrtc/events/Message.ts
+++ b/src/webrtc/events/Message.ts
@@ -7,13 +7,14 @@ const PayloadSchema = {
op: Number,
$d: new Tuple(Object, Number), // or number for heartbeat sequence
$s: Number,
- $t: String
+ $t: String,
};
export async function onMessage(this: WebSocket, buffer: Buffer) {
try {
var data: Payload = JSON.parse(buffer.toString());
- if (data.op !== VoiceOPCodes.IDENTIFY && !this.user_id) return this.close(CLOSECODES.Not_authenticated);
+ if (data.op !== VoiceOPCodes.IDENTIFY && !this.user_id)
+ return this.close(CLOSECODES.Not_authenticated);
// @ts-ignore
const OPCodeHandler = OPCodeHandlers[data.op];
@@ -25,7 +26,11 @@ export async function onMessage(this: WebSocket, buffer: Buffer) {
return;
}
- if (![VoiceOPCodes.HEARTBEAT, VoiceOPCodes.SPEAKING].includes(data.op as VoiceOPCodes)) {
+ if (
+ ![VoiceOPCodes.HEARTBEAT, VoiceOPCodes.SPEAKING].includes(
+ data.op as VoiceOPCodes,
+ )
+ ) {
// @ts-ignore
console.log("[WebRTC] Opcode " + VoiceOPCodes[data.op]);
}
@@ -35,4 +40,4 @@ export async function onMessage(this: WebSocket, buffer: Buffer) {
console.error("[WebRTC] error", error);
// if (!this.CLOSED && this.CLOSING) return this.close(CloseCodes.Unknown_error);
}
-}
\ No newline at end of file
+}
diff --git a/src/webrtc/index.ts b/src/webrtc/index.ts
index 7cecc9b6..ccb088ac 100644
--- a/src/webrtc/index.ts
+++ b/src/webrtc/index.ts
@@ -1,2 +1,2 @@
export * from "./Server";
-export * from "./util/index";
\ No newline at end of file
+export * from "./util/index";
diff --git a/src/webrtc/opcodes/BackendVersion.ts b/src/webrtc/opcodes/BackendVersion.ts
index b4b61c7d..375dd0cc 100644
--- a/src/webrtc/opcodes/BackendVersion.ts
+++ b/src/webrtc/opcodes/BackendVersion.ts
@@ -2,5 +2,8 @@ import { Payload, Send, WebSocket } from "@fosscord/gateway";
import { VoiceOPCodes } from "../util";
export async function onBackendVersion(this: WebSocket, data: Payload) {
- await Send(this, { op: VoiceOPCodes.VOICE_BACKEND_VERSION, d: { voice: "0.8.43", rtc_worker: "0.3.26" } });
-}
\ No newline at end of file
+ await Send(this, {
+ op: VoiceOPCodes.VOICE_BACKEND_VERSION,
+ d: { voice: "0.8.43", rtc_worker: "0.3.26" },
+ });
+}
diff --git a/src/webrtc/opcodes/Heartbeat.ts b/src/webrtc/opcodes/Heartbeat.ts
index 1b6c5bcd..932cd458 100644
--- a/src/webrtc/opcodes/Heartbeat.ts
+++ b/src/webrtc/opcodes/Heartbeat.ts
@@ -1,4 +1,10 @@
-import { CLOSECODES, Payload, Send, setHeartbeat, WebSocket } from "@fosscord/gateway";
+import {
+ CLOSECODES,
+ Payload,
+ Send,
+ setHeartbeat,
+ WebSocket,
+} from "@fosscord/gateway";
import { VoiceOPCodes } from "../util";
export async function onHeartbeat(this: WebSocket, data: Payload) {
@@ -6,4 +12,4 @@ export async function onHeartbeat(this: WebSocket, data: Payload) {
if (isNaN(data.d)) return this.close(CLOSECODES.Decode_error);
await Send(this, { op: VoiceOPCodes.HEARTBEAT_ACK, d: data.d });
-}
\ No newline at end of file
+}
diff --git a/src/webrtc/opcodes/Identify.ts b/src/webrtc/opcodes/Identify.ts
index 19a575ab..45ad6c0a 100644
--- a/src/webrtc/opcodes/Identify.ts
+++ b/src/webrtc/opcodes/Identify.ts
@@ -1,33 +1,46 @@
import { CLOSECODES, Payload, Send, WebSocket } from "@fosscord/gateway";
-import { validateSchema, VoiceIdentifySchema, VoiceState } from "@fosscord/util";
+import {
+ validateSchema,
+ VoiceIdentifySchema,
+ VoiceState,
+} from "@fosscord/util";
import { endpoint, getClients, VoiceOPCodes, PublicIP } from "@fosscord/webrtc";
import SemanticSDP from "semantic-sdp";
const defaultSDP = require("./sdp.json");
export async function onIdentify(this: WebSocket, data: Payload) {
clearTimeout(this.readyTimeout);
- const { server_id, user_id, session_id, token, streams, video } = validateSchema("VoiceIdentifySchema", data.d) as VoiceIdentifySchema;
+ const { server_id, user_id, session_id, token, streams, video } =
+ validateSchema("VoiceIdentifySchema", data.d) as VoiceIdentifySchema;
- const voiceState = await VoiceState.findOne({ where: { guild_id: server_id, user_id, token, session_id } });
+ const voiceState = await VoiceState.findOne({
+ where: { guild_id: server_id, user_id, token, session_id },
+ });
if (!voiceState) return this.close(CLOSECODES.Authentication_failed);
this.user_id = user_id;
this.session_id = session_id;
const sdp = SemanticSDP.SDPInfo.expand(defaultSDP);
- sdp.setDTLS(SemanticSDP.DTLSInfo.expand({ setup: "actpass", hash: "sha-256", fingerprint: endpoint.getDTLSFingerprint() }));
+ sdp.setDTLS(
+ SemanticSDP.DTLSInfo.expand({
+ setup: "actpass",
+ hash: "sha-256",
+ fingerprint: endpoint.getDTLSFingerprint(),
+ }),
+ );
this.client = {
websocket: this,
out: {
- tracks: new Map()
+ tracks: new Map(),
},
in: {
audio_ssrc: 0,
video_ssrc: 0,
- rtx_ssrc: 0
+ rtx_ssrc: 0,
},
sdp,
- channel_id: voiceState.channel_id
+ channel_id: voiceState.channel_id,
};
const clients = getClients(voiceState.channel_id)!;
@@ -51,10 +64,10 @@ export async function onIdentify(this: WebSocket, data: Payload) {
"xsalsa20_poly1305_lite_rtpsize",
"xsalsa20_poly1305_lite",
"xsalsa20_poly1305_suffix",
- "xsalsa20_poly1305"
+ "xsalsa20_poly1305",
],
ip: PublicIP,
- experiments: []
- }
+ experiments: [],
+ },
});
-}
\ No newline at end of file
+}
diff --git a/src/webrtc/opcodes/SelectProtocol.ts b/src/webrtc/opcodes/SelectProtocol.ts
index a3579b34..eadba283 100644
--- a/src/webrtc/opcodes/SelectProtocol.ts
+++ b/src/webrtc/opcodes/SelectProtocol.ts
@@ -6,7 +6,10 @@ import SemanticSDP, { MediaInfo, SDPInfo } from "semantic-sdp";
export async function onSelectProtocol(this: WebSocket, payload: Payload) {
if (!this.client) return;
- const data = validateSchema("SelectProtocolSchema", payload.d) as SelectProtocolSchema;
+ const data = validateSchema(
+ "SelectProtocolSchema",
+ payload.d,
+ ) as SelectProtocolSchema;
const offer = SemanticSDP.SDPInfo.parse("m=audio\n" + data.sdp!);
this.client.sdp!.setICE(offer.getICE());
@@ -25,14 +28,14 @@ export async function onSelectProtocol(this: WebSocket, payload: Payload) {
const candidate = candidates[0];
const answer =
- `m=audio ${port} ICE/SDP`
- + `a=fingerprint:${fingerprint}`
- + `c=IN IP4 ${PublicIP}`
- + `a=rtcp:${port}`
- + `a=ice-ufrag:${ice.getUfrag()}`
- + `a=ice-pwd:${ice.getPwd()}`
- + `a=fingerprint:${fingerprint}`
- + `a=candidate:1 1 ${candidate.getTransport()} ${candidate.getFoundation()} ${candidate.getAddress()} ${candidate.getPort()} typ host`;
+ `m=audio ${port} ICE/SDP` +
+ `a=fingerprint:${fingerprint}` +
+ `c=IN IP4 ${PublicIP}` +
+ `a=rtcp:${port}` +
+ `a=ice-ufrag:${ice.getUfrag()}` +
+ `a=ice-pwd:${ice.getPwd()}` +
+ `a=fingerprint:${fingerprint}` +
+ `a=candidate:1 1 ${candidate.getTransport()} ${candidate.getFoundation()} ${candidate.getAddress()} ${candidate.getPort()} typ host`;
await Send(this, {
op: VoiceOPCodes.SELECT_PROTOCOL_ACK,
@@ -40,7 +43,7 @@ export async function onSelectProtocol(this: WebSocket, payload: Payload) {
video_codec: "H264",
sdp: answer,
media_session_id: this.session_id,
- audio_codec: "opus"
- }
+ audio_codec: "opus",
+ },
});
-}
\ No newline at end of file
+}
diff --git a/src/webrtc/opcodes/Speaking.ts b/src/webrtc/opcodes/Speaking.ts
index e2227040..8488acf8 100644
--- a/src/webrtc/opcodes/Speaking.ts
+++ b/src/webrtc/opcodes/Speaking.ts
@@ -15,8 +15,8 @@ export async function onSpeaking(this: WebSocket, data: Payload) {
d: {
user_id: client.websocket.user_id,
speaking: data.d.speaking,
- ssrc: ssrc?.audio_ssrc || 0
- }
+ ssrc: ssrc?.audio_ssrc || 0,
+ },
});
});
-}
\ No newline at end of file
+}
diff --git a/src/webrtc/opcodes/Video.ts b/src/webrtc/opcodes/Video.ts
index ff20d5a9..dcbc9aa0 100644
--- a/src/webrtc/opcodes/Video.ts
+++ b/src/webrtc/opcodes/Video.ts
@@ -21,8 +21,8 @@ export async function onVideo(this: WebSocket, payload: Payload) {
SemanticSDP.StreamInfo.expand({
id,
// @ts-ignore
- tracks: []
- })
+ tracks: [],
+ }),
);
this.client.in.stream = stream;
@@ -46,8 +46,8 @@ export async function onVideo(this: WebSocket, payload: Payload) {
SemanticSDP.StreamInfo.expand({
id: "out" + this.user_id,
// @ts-ignore
- tracks: []
- })
+ tracks: [],
+ }),
);
this.client.out.stream = out;
@@ -64,20 +64,35 @@ export async function onVideo(this: WebSocket, payload: Payload) {
}
if (d.audio_ssrc) {
- handleSSRC.call(this, "audio", { media: d.audio_ssrc, rtx: d.audio_ssrc + 1 });
+ handleSSRC.call(this, "audio", {
+ media: d.audio_ssrc,
+ rtx: d.audio_ssrc + 1,
+ });
}
if (d.video_ssrc && d.rtx_ssrc) {
- handleSSRC.call(this, "video", { media: d.video_ssrc, rtx: d.rtx_ssrc });
+ handleSSRC.call(this, "video", {
+ media: d.video_ssrc,
+ rtx: d.rtx_ssrc,
+ });
}
}
-function attachTrack(this: WebSocket, track: IncomingStreamTrack, user_id: string) {
+function attachTrack(
+ this: WebSocket,
+ track: IncomingStreamTrack,
+ user_id: string,
+) {
if (!this.client) return;
- const outTrack = this.client.transport!.createOutgoingStreamTrack(track.getMedia());
+ const outTrack = this.client.transport!.createOutgoingStreamTrack(
+ track.getMedia(),
+ );
outTrack.attachTo(track);
this.client.out.stream!.addTrack(outTrack);
var ssrcs = this.client.out.tracks.get(user_id)!;
- if (!ssrcs) ssrcs = this.client.out.tracks.set(user_id, { audio_ssrc: 0, rtx_ssrc: 0, video_ssrc: 0 }).get(user_id)!;
+ if (!ssrcs)
+ ssrcs = this.client.out.tracks
+ .set(user_id, { audio_ssrc: 0, rtx_ssrc: 0, video_ssrc: 0 })
+ .get(user_id)!;
if (track.getMedia() === "audio") {
ssrcs.audio_ssrc = outTrack.getSSRCs().media!;
@@ -90,8 +105,8 @@ function attachTrack(this: WebSocket, track: IncomingStreamTrack, user_id: strin
op: VoiceOPCodes.VIDEO,
d: {
user_id: user_id,
- ...ssrcs
- } as VoiceVideoSchema
+ ...ssrcs,
+ } as VoiceVideoSchema,
});
}
@@ -115,4 +130,4 @@ function handleSSRC(this: WebSocket, type: "audio" | "video", ssrcs: SSRCs) {
attachTrack.call(this, track, client.websocket.user_id);
});
}
-}
\ No newline at end of file
+}
diff --git a/src/webrtc/opcodes/index.ts b/src/webrtc/opcodes/index.ts
index 8c664cce..86e39687 100644
--- a/src/webrtc/opcodes/index.ts
+++ b/src/webrtc/opcodes/index.ts
@@ -15,5 +15,5 @@ export default {
[VoiceOPCodes.VOICE_BACKEND_VERSION]: onBackendVersion,
[VoiceOPCodes.VIDEO]: onVideo,
[VoiceOPCodes.SPEAKING]: onSpeaking,
- [VoiceOPCodes.SELECT_PROTOCOL]: onSelectProtocol
-};
\ No newline at end of file
+ [VoiceOPCodes.SELECT_PROTOCOL]: onSelectProtocol,
+};
diff --git a/src/webrtc/opcodes/sdp.json b/src/webrtc/opcodes/sdp.json
index 4867b9c7..5f7eba38 100644
--- a/src/webrtc/opcodes/sdp.json
+++ b/src/webrtc/opcodes/sdp.json
@@ -417,4 +417,4 @@
}
],
"candidates": []
-}
\ No newline at end of file
+}
diff --git a/src/webrtc/start.ts b/src/webrtc/start.ts
index 9a5f38ee..57361909 100644
--- a/src/webrtc/start.ts
+++ b/src/webrtc/start.ts
@@ -8,6 +8,6 @@ config();
const port = Number(process.env.PORT) || 3004;
const server = new Server({
- port
+ port,
});
-server.start();
\ No newline at end of file
+server.start();
diff --git a/src/webrtc/util/Constants.ts b/src/webrtc/util/Constants.ts
index 64d78e22..d9f1ff60 100644
--- a/src/webrtc/util/Constants.ts
+++ b/src/webrtc/util/Constants.ts
@@ -3,7 +3,7 @@ export enum VoiceStatus {
CONNECTING = 1,
AUTHENTICATING = 2,
RECONNECTING = 3,
- DISCONNECTED = 4
+ DISCONNECTED = 4,
}
export enum VoiceOPCodes {
@@ -22,5 +22,5 @@ export enum VoiceOPCodes {
SESSION_UPDATE = 14,
MEDIA_SINK_WANTS = 15,
VOICE_BACKEND_VERSION = 16,
- CHANNEL_OPTIONS_UPDATE = 17
-}
\ No newline at end of file
+ CHANNEL_OPTIONS_UPDATE = 17,
+}
diff --git a/src/webrtc/util/MediaServer.ts b/src/webrtc/util/MediaServer.ts
index 93230c91..520b8682 100644
--- a/src/webrtc/util/MediaServer.ts
+++ b/src/webrtc/util/MediaServer.ts
@@ -1,5 +1,9 @@
import { WebSocket } from "@fosscord/gateway";
-import MediaServer, { IncomingStream, OutgoingStream, Transport } from "medooze-media-server";
+import MediaServer, {
+ IncomingStream,
+ OutgoingStream,
+ Transport,
+} from "medooze-media-server";
import SemanticSDP from "semantic-sdp";
MediaServer.enableLog(true);
@@ -13,7 +17,11 @@ try {
MediaServer.setPortRange(min, max);
} catch (error) {
- console.error("Invalid env var: WEBRTC_PORT_RANGE", process.env.WEBRTC_PORT_RANGE, error);
+ console.error(
+ "Invalid env var: WEBRTC_PORT_RANGE",
+ process.env.WEBRTC_PORT_RANGE,
+ error,
+ );
process.exit(1);
}
@@ -48,4 +56,4 @@ export interface Client {
export function getClients(channel_id: string) {
if (!channels.has(channel_id)) channels.set(channel_id, new Set());
return channels.get(channel_id)!;
-}
\ No newline at end of file
+}
diff --git a/src/webrtc/util/index.ts b/src/webrtc/util/index.ts
index 2e09bc48..f0d49049 100644
--- a/src/webrtc/util/index.ts
+++ b/src/webrtc/util/index.ts
@@ -1,2 +1,2 @@
export * from "./Constants";
-export * from "./MediaServer";
\ No newline at end of file
+export * from "./MediaServer";
|
