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authorMadeline <46743919+MaddyUnderStars@users.noreply.github.com>2022-01-21 21:04:45 +1100
committerMadeline <46743919+MaddyUnderStars@users.noreply.github.com>2022-01-21 21:04:45 +1100
commit4847351daa226cbd71fc8676b6be7516c6e76253 (patch)
treef10959c0374e8967a79eb05310944580fd5c18ca /webrtc/src/opcodes
parentmore fuckery (diff)
downloadserver-4847351daa226cbd71fc8676b6be7516c6e76253.tar.xz
mmmm
Diffstat (limited to 'webrtc/src/opcodes')
-rw-r--r--webrtc/src/opcodes/Identify.ts79
-rw-r--r--webrtc/src/opcodes/SelectProtocol.ts37
-rw-r--r--webrtc/src/opcodes/index.ts2
3 files changed, 67 insertions, 51 deletions
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