diff --git a/webrtc/src/Server.ts b/webrtc/src/Server.ts
deleted file mode 100644
index 32b795ea..00000000
--- a/webrtc/src/Server.ts
+++ /dev/null
@@ -1,56 +0,0 @@
-import { closeDatabase, Config, initDatabase, initEvent } from "@fosscord/util";
-import dotenv from "dotenv";
-import http from "http";
-import ws from "ws";
-import { Connection } from "./events/Connection";
-dotenv.config();
-
-export class Server {
- public ws: ws.Server;
- public port: number;
- public server: http.Server;
- public production: boolean;
-
- constructor({ port, server, production }: { port: number; server?: http.Server; production?: boolean }) {
- this.port = port;
- this.production = production || false;
-
- if (server) this.server = server;
- else {
- this.server = http.createServer(function (req, res) {
- res.writeHead(200).end("Online");
- });
- }
-
- // this.server.on("upgrade", (request, socket, head) => {
- // if (!request.url?.includes("voice")) return;
- // this.ws.handleUpgrade(request, socket, head, (socket) => {
- // // @ts-ignore
- // socket.server = this;
- // this.ws.emit("connection", socket, request);
- // });
- // });
-
- this.ws = new ws.Server({
- maxPayload: 1024 * 1024 * 100,
- server: this.server,
- });
- this.ws.on("connection", Connection);
- this.ws.on("error", console.error);
- }
-
- async start(): Promise<void> {
- await initDatabase();
- await Config.init();
- await initEvent();
- if (!this.server.listening) {
- this.server.listen(this.port);
- console.log(`[WebRTC] online on 0.0.0.0:${this.port}`);
- }
- }
-
- async stop() {
- closeDatabase();
- this.server.close();
- }
-}
\ No newline at end of file
diff --git a/webrtc/src/events/Close.ts b/webrtc/src/events/Close.ts
deleted file mode 100644
index 1c203653..00000000
--- a/webrtc/src/events/Close.ts
+++ /dev/null
@@ -1,9 +0,0 @@
-import { WebSocket } from "@fosscord/gateway";
-import { Session } from "@fosscord/util";
-
-export async function onClose(this: WebSocket, code: number, reason: string) {
- console.log("[WebRTC] closed", code, reason.toString());
-
- if (this.session_id) await Session.delete({ session_id: this.session_id });
- this.removeAllListeners();
-}
\ No newline at end of file
diff --git a/webrtc/src/events/Connection.ts b/webrtc/src/events/Connection.ts
deleted file mode 100644
index bf228d64..00000000
--- a/webrtc/src/events/Connection.ts
+++ /dev/null
@@ -1,60 +0,0 @@
-import { CLOSECODES, Send, setHeartbeat, WebSocket } from "@fosscord/gateway";
-import { IncomingMessage } from "http";
-import { URL } from "url";
-import WS from "ws";
-import { VoiceOPCodes } from "../util";
-import { onClose } from "./Close";
-import { onMessage } from "./Message";
-var erlpack: any;
-try {
- erlpack = require("@yukikaze-bot/erlpack");
-} catch (error) {}
-
-// TODO: check rate limit
-// TODO: specify rate limit in config
-// TODO: check msg max size
-
-export async function Connection(this: WS.Server, socket: WebSocket, request: IncomingMessage) {
- try {
- socket.on("close", onClose.bind(socket));
- socket.on("message", onMessage.bind(socket));
- console.log("[WebRTC] new connection", request.url);
-
- if (process.env.WS_LOGEVENTS) {
- [
- "close",
- "error",
- "upgrade",
- //"message",
- "open",
- "ping",
- "pong",
- "unexpected-response"
- ].forEach((x) => {
- socket.on(x, (y) => console.log("[WebRTC]", x, y));
- });
- }
-
- const { searchParams } = new URL(`http://localhost${request.url}`);
-
- socket.encoding = "json";
- socket.version = Number(searchParams.get("v")) || 5;
- if (socket.version < 3) return socket.close(CLOSECODES.Unknown_error, "invalid version");
-
- setHeartbeat(socket);
-
- socket.readyTimeout = setTimeout(() => {
- return socket.close(CLOSECODES.Session_timed_out);
- }, 1000 * 30);
-
- await Send(socket, {
- op: VoiceOPCodes.HELLO,
- d: {
- 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/webrtc/src/events/Message.ts b/webrtc/src/events/Message.ts
deleted file mode 100644
index 8f75a815..00000000
--- a/webrtc/src/events/Message.ts
+++ /dev/null
@@ -1,38 +0,0 @@
-import { CLOSECODES, Payload, WebSocket } from "@fosscord/gateway";
-import { Tuple } from "lambert-server";
-import OPCodeHandlers from "../opcodes";
-import { VoiceOPCodes } from "../util";
-
-const PayloadSchema = {
- op: Number,
- $d: new Tuple(Object, Number), // or number for heartbeat sequence
- $s: Number,
- $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);
-
- // @ts-ignore
- const OPCodeHandler = OPCodeHandlers[data.op];
- if (!OPCodeHandler) {
- // @ts-ignore
- console.error("[WebRTC] Unkown opcode " + VoiceOPCodes[data.op]);
- // TODO: if all opcodes are implemented comment this out:
- // this.close(CloseCodes.Unknown_opcode);
- return;
- }
-
- if (![VoiceOPCodes.HEARTBEAT, VoiceOPCodes.SPEAKING].includes(data.op as VoiceOPCodes)) {
- // @ts-ignore
- console.log("[WebRTC] Opcode " + VoiceOPCodes[data.op]);
- }
-
- return await OPCodeHandler.call(this, data);
- } catch (error) {
- 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/webrtc/src/index.ts b/webrtc/src/index.ts
deleted file mode 100644
index 7cecc9b6..00000000
--- a/webrtc/src/index.ts
+++ /dev/null
@@ -1,2 +0,0 @@
-export * from "./Server";
-export * from "./util/index";
\ No newline at end of file
diff --git a/webrtc/src/opcodes/BackendVersion.ts b/webrtc/src/opcodes/BackendVersion.ts
deleted file mode 100644
index b4b61c7d..00000000
--- a/webrtc/src/opcodes/BackendVersion.ts
+++ /dev/null
@@ -1,6 +0,0 @@
-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
diff --git a/webrtc/src/opcodes/Heartbeat.ts b/webrtc/src/opcodes/Heartbeat.ts
deleted file mode 100644
index 1b6c5bcd..00000000
--- a/webrtc/src/opcodes/Heartbeat.ts
+++ /dev/null
@@ -1,9 +0,0 @@
-import { CLOSECODES, Payload, Send, setHeartbeat, WebSocket } from "@fosscord/gateway";
-import { VoiceOPCodes } from "../util";
-
-export async function onHeartbeat(this: WebSocket, data: Payload) {
- setHeartbeat(this);
- 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/webrtc/src/opcodes/Identify.ts b/webrtc/src/opcodes/Identify.ts
deleted file mode 100644
index 19a575ab..00000000
--- a/webrtc/src/opcodes/Identify.ts
+++ /dev/null
@@ -1,60 +0,0 @@
-import { CLOSECODES, Payload, Send, WebSocket } from "@fosscord/gateway";
-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 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() }));
-
- this.client = {
- websocket: this,
- out: {
- tracks: new Map()
- },
- in: {
- audio_ssrc: 0,
- video_ssrc: 0,
- rtx_ssrc: 0
- },
- sdp,
- channel_id: voiceState.channel_id
- };
-
- const clients = getClients(voiceState.channel_id)!;
- clients.add(this.client);
-
- this.on("close", () => {
- clients.delete(this.client!);
- });
-
- await Send(this, {
- op: VoiceOPCodes.READY,
- d: {
- streams: [
- // { type: "video", ssrc: this.ssrc + 1, rtx_ssrc: this.ssrc + 2, rid: "100", quality: 100, active: false }
- ],
- ssrc: -1,
- port: endpoint.getLocalPort(),
- modes: [
- "aead_aes256_gcm_rtpsize",
- "aead_aes256_gcm",
- "xsalsa20_poly1305_lite_rtpsize",
- "xsalsa20_poly1305_lite",
- "xsalsa20_poly1305_suffix",
- "xsalsa20_poly1305"
- ],
- ip: PublicIP,
- experiments: []
- }
- });
-}
\ No newline at end of file
diff --git a/webrtc/src/opcodes/SelectProtocol.ts b/webrtc/src/opcodes/SelectProtocol.ts
deleted file mode 100644
index a3579b34..00000000
--- a/webrtc/src/opcodes/SelectProtocol.ts
+++ /dev/null
@@ -1,46 +0,0 @@
-import { Payload, Send, WebSocket } from "@fosscord/gateway";
-import { SelectProtocolSchema, validateSchema } from "@fosscord/util";
-import { endpoint, PublicIP, VoiceOPCodes } from "@fosscord/webrtc";
-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 offer = SemanticSDP.SDPInfo.parse("m=audio\n" + data.sdp!);
- this.client.sdp!.setICE(offer.getICE());
- this.client.sdp!.setDTLS(offer.getDTLS());
-
- const transport = endpoint.createTransport(this.client.sdp!);
- this.client.transport = transport;
- transport.setRemoteProperties(this.client.sdp!);
- transport.setLocalProperties(this.client.sdp!);
-
- const dtls = transport.getLocalDTLSInfo();
- const ice = transport.getLocalICEInfo();
- const port = endpoint.getLocalPort();
- const fingerprint = dtls.getHash() + " " + dtls.getFingerprint();
- const candidates = transport.getLocalCandidates();
- 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`;
-
- await Send(this, {
- op: VoiceOPCodes.SELECT_PROTOCOL_ACK,
- d: {
- video_codec: "H264",
- sdp: answer,
- media_session_id: this.session_id,
- audio_codec: "opus"
- }
- });
-}
\ No newline at end of file
diff --git a/webrtc/src/opcodes/Speaking.ts b/webrtc/src/opcodes/Speaking.ts
deleted file mode 100644
index e2227040..00000000
--- a/webrtc/src/opcodes/Speaking.ts
+++ /dev/null
@@ -1,22 +0,0 @@
-import { Payload, Send, WebSocket } from "@fosscord/gateway";
-import { getClients, VoiceOPCodes } from "../util";
-
-// {"speaking":1,"delay":5,"ssrc":2805246727}
-
-export async function onSpeaking(this: WebSocket, data: Payload) {
- if (!this.client) return;
-
- getClients(this.client.channel_id).forEach((client) => {
- if (client === this.client) return;
- const ssrc = this.client!.out.tracks.get(client.websocket.user_id);
-
- Send(client.websocket, {
- op: VoiceOPCodes.SPEAKING,
- d: {
- user_id: client.websocket.user_id,
- speaking: data.d.speaking,
- ssrc: ssrc?.audio_ssrc || 0
- }
- });
- });
-}
\ No newline at end of file
diff --git a/webrtc/src/opcodes/Video.ts b/webrtc/src/opcodes/Video.ts
deleted file mode 100644
index ff20d5a9..00000000
--- a/webrtc/src/opcodes/Video.ts
+++ /dev/null
@@ -1,118 +0,0 @@
-import { Payload, Send, WebSocket } from "@fosscord/gateway";
-import { validateSchema, VoiceVideoSchema } from "@fosscord/util";
-import { channels, getClients, VoiceOPCodes } from "@fosscord/webrtc";
-import { IncomingStreamTrack, SSRCs } from "medooze-media-server";
-import SemanticSDP from "semantic-sdp";
-
-export async function onVideo(this: WebSocket, payload: Payload) {
- if (!this.client) return;
- const { transport, channel_id } = this.client;
- if (!transport) return;
- const d = validateSchema("VoiceVideoSchema", payload.d) as VoiceVideoSchema;
-
- await Send(this, { op: VoiceOPCodes.MEDIA_SINK_WANTS, d: { any: 100 } });
-
- const id = "stream" + this.user_id;
-
- var stream = this.client.in.stream!;
- if (!stream) {
- stream = this.client.transport!.createIncomingStream(
- // @ts-ignore
- SemanticSDP.StreamInfo.expand({
- id,
- // @ts-ignore
- tracks: []
- })
- );
- this.client.in.stream = stream;
-
- const interval = setInterval(() => {
- for (const track of stream.getTracks()) {
- for (const layer of Object.values(track.getStats())) {
- console.log(track.getId(), layer.total);
- }
- }
- }, 5000);
-
- stream.on("stopped", () => {
- console.log("stream stopped");
- clearInterval(interval);
- });
- this.on("close", () => {
- transport!.stop();
- });
- const out = transport.createOutgoingStream(
- // @ts-ignore
- SemanticSDP.StreamInfo.expand({
- id: "out" + this.user_id,
- // @ts-ignore
- tracks: []
- })
- );
- this.client.out.stream = out;
-
- const clients = channels.get(channel_id)!;
-
- clients.forEach((client) => {
- if (client.websocket.user_id === this.user_id) return;
- if (!client.in.stream) return;
-
- client.in.stream?.getTracks().forEach((track) => {
- attachTrack.call(this, track, client.websocket.user_id);
- });
- });
- }
-
- if (d.audio_ssrc) {
- 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 });
- }
-}
-
-function attachTrack(this: WebSocket, track: IncomingStreamTrack, user_id: string) {
- if (!this.client) return;
- 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 (track.getMedia() === "audio") {
- ssrcs.audio_ssrc = outTrack.getSSRCs().media!;
- } else if (track.getMedia() === "video") {
- ssrcs.video_ssrc = outTrack.getSSRCs().media!;
- ssrcs.rtx_ssrc = outTrack.getSSRCs().rtx!;
- }
-
- Send(this, {
- op: VoiceOPCodes.VIDEO,
- d: {
- user_id: user_id,
- ...ssrcs
- } as VoiceVideoSchema
- });
-}
-
-function handleSSRC(this: WebSocket, type: "audio" | "video", ssrcs: SSRCs) {
- if (!this.client) return;
- const stream = this.client.in.stream!;
- const transport = this.client.transport!;
-
- const id = type + ssrcs.media;
- var track = stream.getTrack(id);
- if (!track) {
- console.log("createIncomingStreamTrack", id);
- track = transport.createIncomingStreamTrack(type, { id, ssrcs });
- stream.addTrack(track);
-
- const clients = getClients(this.client.channel_id)!;
- clients.forEach((client) => {
- if (client.websocket.user_id === this.user_id) return;
- if (!client.out.stream) return;
-
- attachTrack.call(this, track, client.websocket.user_id);
- });
- }
-}
\ No newline at end of file
diff --git a/webrtc/src/opcodes/index.ts b/webrtc/src/opcodes/index.ts
deleted file mode 100644
index 8c664cce..00000000
--- a/webrtc/src/opcodes/index.ts
+++ /dev/null
@@ -1,19 +0,0 @@
-import { Payload, WebSocket } from "@fosscord/gateway";
-import { VoiceOPCodes } from "../util";
-import { onBackendVersion } from "./BackendVersion";
-import { onHeartbeat } from "./Heartbeat";
-import { onIdentify } from "./Identify";
-import { onSelectProtocol } from "./SelectProtocol";
-import { onSpeaking } from "./Speaking";
-import { onVideo } from "./Video";
-
-export type OPCodeHandler = (this: WebSocket, data: Payload) => any;
-
-export default {
- [VoiceOPCodes.HEARTBEAT]: onHeartbeat,
- [VoiceOPCodes.IDENTIFY]: onIdentify,
- [VoiceOPCodes.VOICE_BACKEND_VERSION]: onBackendVersion,
- [VoiceOPCodes.VIDEO]: onVideo,
- [VoiceOPCodes.SPEAKING]: onSpeaking,
- [VoiceOPCodes.SELECT_PROTOCOL]: onSelectProtocol
-};
\ No newline at end of file
diff --git a/webrtc/src/opcodes/sdp.json b/webrtc/src/opcodes/sdp.json
deleted file mode 100644
index 4867b9c7..00000000
--- a/webrtc/src/opcodes/sdp.json
+++ /dev/null
@@ -1,420 +0,0 @@
-{
- "version": 0,
- "streams": [],
- "medias": [
- {
- "id": "0",
- "type": "audio",
- "direction": "sendrecv",
- "codecs": [
- {
- "codec": "opus",
- "type": 111,
- "channels": 2,
- "params": {
- "minptime": "10",
- "useinbandfec": "1"
- },
- "rtcpfbs": [
- {
- "id": "transport-cc"
- }
- ]
- }
- ],
- "extensions": {
- "1": "urn:ietf:params:rtp-hdrext:ssrc-audio-level",
- "2": "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
- "3": "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
- "4": "urn:ietf:params:rtp-hdrext:sdes:mid"
- }
- },
- {
- "id": "1",
- "type": "video",
- "direction": "sendrecv",
- "codecs": [
- {
- "codec": "VP8",
- "type": 96,
- "rtx": 97,
- "rtcpfbs": [
- {
- "id": "goog-remb"
- },
- {
- "id": "transport-cc"
- },
- {
- "id": "ccm",
- "params": ["fir"]
- },
- {
- "id": "nack"
- },
- {
- "id": "nack",
- "params": ["pli"]
- }
- ]
- },
- {
- "codec": "VP9",
- "type": 98,
- "rtx": 99,
- "params": {
- "profile-id": "0"
- },
- "rtcpfbs": [
- {
- "id": "goog-remb"
- },
- {
- "id": "transport-cc"
- },
- {
- "id": "ccm",
- "params": ["fir"]
- },
- {
- "id": "nack"
- },
- {
- "id": "nack",
- "params": ["pli"]
- }
- ]
- },
- {
- "codec": "VP9",
- "type": 100,
- "rtx": 101,
- "params": {
- "profile-id": "2"
- },
- "rtcpfbs": [
- {
- "id": "goog-remb"
- },
- {
- "id": "transport-cc"
- },
- {
- "id": "ccm",
- "params": ["fir"]
- },
- {
- "id": "nack"
- },
- {
- "id": "nack",
- "params": ["pli"]
- }
- ]
- },
- {
- "codec": "VP9",
- "type": 102,
- "rtx": 122,
- "params": {
- "profile-id": "1"
- },
- "rtcpfbs": [
- {
- "id": "goog-remb"
- },
- {
- "id": "transport-cc"
- },
- {
- "id": "ccm",
- "params": ["fir"]
- },
- {
- "id": "nack"
- },
- {
- "id": "nack",
- "params": ["pli"]
- }
- ]
- },
- {
- "codec": "H264",
- "type": 127,
- "rtx": 121,
- "params": {
- "level-asymmetry-allowed": "1",
- "packetization-mode": "1",
- "profile-level-id": "42001f"
- },
- "rtcpfbs": [
- {
- "id": "goog-remb"
- },
- {
- "id": "transport-cc"
- },
- {
- "id": "ccm",
- "params": ["fir"]
- },
- {
- "id": "nack"
- },
- {
- "id": "nack",
- "params": ["pli"]
- }
- ]
- },
- {
- "codec": "H264",
- "type": 125,
- "rtx": 107,
- "params": {
- "level-asymmetry-allowed": "1",
- "packetization-mode": "0",
- "profile-level-id": "42001f"
- },
- "rtcpfbs": [
- {
- "id": "goog-remb"
- },
- {
- "id": "transport-cc"
- },
- {
- "id": "ccm",
- "params": ["fir"]
- },
- {
- "id": "nack"
- },
- {
- "id": "nack",
- "params": ["pli"]
- }
- ]
- },
- {
- "codec": "H264",
- "type": 108,
- "rtx": 109,
- "params": {
- "level-asymmetry-allowed": "1",
- "packetization-mode": "1",
- "profile-level-id": "42e01f"
- },
- "rtcpfbs": [
- {
- "id": "goog-remb"
- },
- {
- "id": "transport-cc"
- },
- {
- "id": "ccm",
- "params": ["fir"]
- },
- {
- "id": "nack"
- },
- {
- "id": "nack",
- "params": ["pli"]
- }
- ]
- },
- {
- "codec": "H264",
- "type": 124,
- "rtx": 120,
- "params": {
- "level-asymmetry-allowed": "1",
- "packetization-mode": "0",
- "profile-level-id": "42e01f"
- },
- "rtcpfbs": [
- {
- "id": "goog-remb"
- },
- {
- "id": "transport-cc"
- },
- {
- "id": "ccm",
- "params": ["fir"]
- },
- {
- "id": "nack"
- },
- {
- "id": "nack",
- "params": ["pli"]
- }
- ]
- },
- {
- "codec": "H264",
- "type": 123,
- "rtx": 119,
- "params": {
- "level-asymmetry-allowed": "1",
- "packetization-mode": "1",
- "profile-level-id": "4d001f"
- },
- "rtcpfbs": [
- {
- "id": "goog-remb"
- },
- {
- "id": "transport-cc"
- },
- {
- "id": "ccm",
- "params": ["fir"]
- },
- {
- "id": "nack"
- },
- {
- "id": "nack",
- "params": ["pli"]
- }
- ]
- },
- {
- "codec": "H264",
- "type": 35,
- "rtx": 36,
- "params": {
- "level-asymmetry-allowed": "1",
- "packetization-mode": "0",
- "profile-level-id": "4d001f"
- },
- "rtcpfbs": [
- {
- "id": "goog-remb"
- },
- {
- "id": "transport-cc"
- },
- {
- "id": "ccm",
- "params": ["fir"]
- },
- {
- "id": "nack"
- },
- {
- "id": "nack",
- "params": ["pli"]
- }
- ]
- },
- {
- "codec": "H264",
- "type": 37,
- "rtx": 38,
- "params": {
- "level-asymmetry-allowed": "1",
- "packetization-mode": "1",
- "profile-level-id": "f4001f"
- },
- "rtcpfbs": [
- {
- "id": "goog-remb"
- },
- {
- "id": "transport-cc"
- },
- {
- "id": "ccm",
- "params": ["fir"]
- },
- {
- "id": "nack"
- },
- {
- "id": "nack",
- "params": ["pli"]
- }
- ]
- },
- {
- "codec": "H264",
- "type": 39,
- "rtx": 40,
- "params": {
- "level-asymmetry-allowed": "1",
- "packetization-mode": "0",
- "profile-level-id": "f4001f"
- },
- "rtcpfbs": [
- {
- "id": "goog-remb"
- },
- {
- "id": "transport-cc"
- },
- {
- "id": "ccm",
- "params": ["fir"]
- },
- {
- "id": "nack"
- },
- {
- "id": "nack",
- "params": ["pli"]
- }
- ]
- },
- {
- "codec": "H264",
- "type": 114,
- "rtx": 115,
- "params": {
- "level-asymmetry-allowed": "1",
- "packetization-mode": "1",
- "profile-level-id": "64001f"
- },
- "rtcpfbs": [
- {
- "id": "goog-remb"
- },
- {
- "id": "transport-cc"
- },
- {
- "id": "ccm",
- "params": ["fir"]
- },
- {
- "id": "nack"
- },
- {
- "id": "nack",
- "params": ["pli"]
- }
- ]
- }
- ],
- "extensions": {
- "2": "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
- "3": "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
- "4": "urn:ietf:params:rtp-hdrext:sdes:mid",
- "5": "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay",
- "6": "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type",
- "7": "http://www.webrtc.org/experiments/rtp-hdrext/video-timing",
- "8": "http://www.webrtc.org/experiments/rtp-hdrext/color-space",
- "10": "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id",
- "11": "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id",
- "13": "urn:3gpp:video-orientation",
- "14": "urn:ietf:params:rtp-hdrext:toffset"
- }
- }
- ],
- "candidates": []
-}
\ No newline at end of file
diff --git a/webrtc/src/start.ts b/webrtc/src/start.ts
deleted file mode 100644
index 9a5f38ee..00000000
--- a/webrtc/src/start.ts
+++ /dev/null
@@ -1,13 +0,0 @@
-process.on("uncaughtException", console.error);
-process.on("unhandledRejection", console.error);
-
-import { config } from "dotenv";
-import { Server } from "./Server";
-config();
-
-const port = Number(process.env.PORT) || 3004;
-
-const server = new Server({
- port
-});
-server.start();
\ No newline at end of file
diff --git a/webrtc/src/util/Constants.ts b/webrtc/src/util/Constants.ts
deleted file mode 100644
index 64d78e22..00000000
--- a/webrtc/src/util/Constants.ts
+++ /dev/null
@@ -1,26 +0,0 @@
-export enum VoiceStatus {
- CONNECTED = 0,
- CONNECTING = 1,
- AUTHENTICATING = 2,
- RECONNECTING = 3,
- DISCONNECTED = 4
-}
-
-export enum VoiceOPCodes {
- IDENTIFY = 0,
- SELECT_PROTOCOL = 1,
- READY = 2,
- HEARTBEAT = 3,
- SELECT_PROTOCOL_ACK = 4,
- SPEAKING = 5,
- HEARTBEAT_ACK = 6,
- RESUME = 7,
- HELLO = 8,
- RESUMED = 9,
- VIDEO = 12,
- CLIENT_DISCONNECT = 13,
- SESSION_UPDATE = 14,
- MEDIA_SINK_WANTS = 15,
- VOICE_BACKEND_VERSION = 16,
- CHANNEL_OPTIONS_UPDATE = 17
-}
\ No newline at end of file
diff --git a/webrtc/src/util/MediaServer.ts b/webrtc/src/util/MediaServer.ts
deleted file mode 100644
index 93230c91..00000000
--- a/webrtc/src/util/MediaServer.ts
+++ /dev/null
@@ -1,51 +0,0 @@
-import { WebSocket } from "@fosscord/gateway";
-import MediaServer, { IncomingStream, OutgoingStream, Transport } from "medooze-media-server";
-import SemanticSDP from "semantic-sdp";
-MediaServer.enableLog(true);
-
-export const PublicIP = process.env.PUBLIC_IP || "127.0.0.1";
-
-try {
- const range = process.env.WEBRTC_PORT_RANGE || "4000";
- var ports = range.split("-");
- const min = Number(ports[0]);
- const max = Number(ports[1]);
-
- MediaServer.setPortRange(min, max);
-} catch (error) {
- console.error("Invalid env var: WEBRTC_PORT_RANGE", process.env.WEBRTC_PORT_RANGE, error);
- process.exit(1);
-}
-
-export const endpoint = MediaServer.createEndpoint(PublicIP);
-
-export const channels = new Map<string, Set<Client>>();
-
-export interface Client {
- transport?: Transport;
- websocket: WebSocket;
- out: {
- stream?: OutgoingStream;
- tracks: Map<
- string,
- {
- audio_ssrc: number;
- video_ssrc: number;
- rtx_ssrc: number;
- }
- >;
- };
- in: {
- stream?: IncomingStream;
- audio_ssrc: number;
- video_ssrc: number;
- rtx_ssrc: number;
- };
- sdp: SemanticSDP.SDPInfo;
- channel_id: string;
-}
-
-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/webrtc/src/util/index.ts b/webrtc/src/util/index.ts
deleted file mode 100644
index 2e09bc48..00000000
--- a/webrtc/src/util/index.ts
+++ /dev/null
@@ -1,2 +0,0 @@
-export * from "./Constants";
-export * from "./MediaServer";
\ No newline at end of file
|
