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
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