diff --git a/webrtc/src/opcodes/Identify.ts b/webrtc/src/opcodes/Identify.ts
index 6bbed04c..c31870c8 100644
--- a/webrtc/src/opcodes/Identify.ts
+++ b/webrtc/src/opcodes/Identify.ts
@@ -2,10 +2,6 @@ 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";
-
-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({
@@ -15,52 +11,31 @@ export async function onIdentify(this: Server, socket: WebSocket, data: Payload)
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,
- }))
- }
- });
+ /*
+ //discord proper sends:
+ {
+ "streams": [
+ { "type": "video", "ssrc": 1311885, "rtx_ssrc": 1311886, "rid": "50", "quality": 50, "active": false },
+ { "type": "video", "ssrc": 1311887, "rtx_ssrc": 1311888, "rid": "100", "quality": 100, "active": false }
+ ],
+ "ssrc": 1311884,
+ "port": 50008,
+ "modes": [
+ "aead_aes256_gcm_rtpsize",
+ "aead_aes256_gcm",
+ "xsalsa20_poly1305_lite_rtpsize",
+ "xsalsa20_poly1305_lite",
+ "xsalsa20_poly1305_suffix",
+ "xsalsa20_poly1305"
+ ],
+ "ip": "109.200.214.158",
+ "experiments": [
+ "bwe_conservative_link_estimate",
+ "bwe_remote_locus_client",
+ "fixed_keyframe_interval"
+ ]
+ }
+ */
socket.send(JSON.stringify({
op: VoiceOPCodes.READY,
@@ -71,11 +46,13 @@ export async function onIdentify(this: Server, socket: WebSocket, data: Payload)
//@ts-ignore
port: transport.iceCandidates[0].port,
modes: [
- "xsalsa20_poly1305",
+ "aead_aes256_gcm_rtpsize",
+ // "xsalsa20_poly1305",
// "xsalsa20_poly1305_suffix",
// "xsalsa20_poly1305_lite",
],
heartbeat_interval: 1,
+ experiments: [],
},
}));
}
\ No newline at end of file
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: {
diff --git a/webrtc/src/opcodes/index.ts b/webrtc/src/opcodes/index.ts
index 2fe69c38..36d30e7d 100644
--- a/webrtc/src/opcodes/index.ts
+++ b/webrtc/src/opcodes/index.ts
@@ -32,4 +32,6 @@ export default {
//op 11?
[VoiceOPCodes.CLIENT_CONNECT]: onConnect, //op 12
//op 13?
+ //op 15?
+ //op 16? empty data on client send but server sends {"voice":"0.8.24+bugfix.voice.streams.opt.branch-ffcefaff7","rtc_worker":"0.3.14-crypto-collision-copy"}
};
\ No newline at end of file
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