diff --git a/webrtc/src/opcodes/SelectProtocol.ts b/webrtc/src/opcodes/SelectProtocol.ts
index 24e8ef5f..08335ade 100644
--- a/webrtc/src/opcodes/SelectProtocol.ts
+++ b/webrtc/src/opcodes/SelectProtocol.ts
@@ -2,6 +2,9 @@ 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';
/*
{
@@ -66,6 +69,40 @@ import { Server } from "../Server";
*/
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}`,
+ })),
+ */
+
+ 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,
+ });
+
socket.send(JSON.stringify({
op: VoiceOPCodes.SESSION_DESCRIPTION,
d: {
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