diff --git a/webrtc/src/opcodes/SelectProtocol.ts b/webrtc/src/opcodes/SelectProtocol.ts
new file mode 100644
index 00000000..36527a8b
--- /dev/null
+++ b/webrtc/src/opcodes/SelectProtocol.ts
@@ -0,0 +1,128 @@
+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';
+
+/*
+ {
+ 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",
+ },
+ }
+*/
+
+var test_hasMadeProducer = false;
+
+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}`,
+ })),
+ */
+
+ if (!test_hasMadeProducer) {
+ 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,
+ });
+
+ const consumer = await transport.consume({
+ producerId: producer.id,
+ paused: false,
+ rtpCapabilities,
+ })
+
+ test_hasMadeProducer = true;
+ }
+
+ socket.send(JSON.stringify({
+ op: VoiceOPCodes.SESSION_DESCRIPTION,
+ d: {
+ 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: "xsalsa20_poly1305",
+ media_session_id: this.mediasoupTransports[0].id,
+ audio_codec: data.d.codecs.find((x: any) => x.type === "audio").name,
+ }
+ }));
+}
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