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This commit is contained in:
Madeline
2023-02-09 23:34:32 +11:00
parent c170af41ce
commit e849d24a7a
2 changed files with 18 additions and 50 deletions
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src/webrtc/opcodes/Identify.ts
+8 -8
View File
@@ -73,14 +73,14 @@ export async function onIdentify(this: WebSocket, data: Payload) {
op: VoiceOPCodes.READY,
d: {
streams: [
{
type: "video",
ssrc: this.webrtcClient.in.video_ssrc + 1,
rtx_ssrc: this.webrtcClient.in.video_ssrc + 2,
rid: "100",
quality: 100,
active: false,
},
// {
// type: "video",
// ssrc: this.webrtcClient.in.video_ssrc + 1,
// rtx_ssrc: this.webrtcClient.in.video_ssrc + 2,
// rid: "100",
// quality: 100,
// active: false,
// },
],
ssrc: this.webrtcClient.in.audio_ssrc,
port: endpoint.getLocalPort(),
src/webrtc/opcodes/SelectProtocol.ts
+10 -42
View File
@@ -19,6 +19,7 @@
import { Payload, Send, WebSocket } from "@fosscord/gateway";
import { SelectProtocolSchema, validateSchema } from "@fosscord/util";
import { endpoint, PublicIP, VoiceOPCodes } from "@fosscord/webrtc";
import MediaServer from "medooze-media-server";
import SemanticSDP, { MediaInfo } from "semantic-sdp";
import DefaultSDP from "./sdp.json";
@@ -61,48 +62,15 @@ export async function onSelectProtocol(this: WebSocket, payload: Payload) {
a=candidate:1 1 UDP 4261412862 66.22.206.174 50026 typ host\n
*/
const answer = offer.answer({
dtls: dtls,
ice: ice,
candidates: endpoint.getLocalCandidates(),
capabilities: {
audio: {
codecs: ["opus"],
rtx: true,
rtcpfbs: [{ id: "transport-cc" }],
extensions: [
"urn:ietf:params:rtp-hdrext:ssrc-audio-level",
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
"urn:ietf:params:rtp-hdrext:sdes:mid",
],
},
},
});
// the Video handler creates streams but we need streams now so idk
for (const offered of offer.getStreams().values()) {
const incomingStream = transport.createIncomingStream(offered);
const outgoingStream = transport.createOutgoingStream({
audio: true,
});
outgoingStream.attachTo(incomingStream);
this.webrtcClient.in.stream = incomingStream;
this.webrtcClient.out.stream = outgoingStream;
const info = outgoingStream.getStreamInfo();
answer.addStream(info);
}
// const answer =
// `m=audio ${port} ICE/SDP\n` +
// `a=fingerprint:${fingerprint}\n` +
// `c=IN IP4 ${PublicIP}\n` +
// `a=rtcp:${port}\n` +
// `a=ice-ufrag:${ice.getUfrag()}\n` +
// `a=ice-pwd:${ice.getPwd()}\n` +
// `a=fingerprint:${fingerprint}\n` +
// `a=candidate:1 1 ${candidate.getTransport()} ${candidate.getFoundation()} ${candidate.getAddress()} ${candidate.getPort()} typ host\n`;
const answer =
`m=audio ${port} ICE/SDP\n` +
`a=fingerprint:${fingerprint}\n` +
`c=IN IP4 ${PublicIP}\n` +
`a=rtcp:${port}\n` +
`a=ice-ufrag:${ice.getUfrag()}\n` +
`a=ice-pwd:${ice.getPwd()}\n` +
`a=fingerprint:${fingerprint}\n` +
`a=candidate:1 1 ${candidate.getTransport()} ${candidate.getFoundation()} ${candidate.getAddress()} ${candidate.getPort()} typ host\n`;
await Send(this, {
op: VoiceOPCodes.SELECT_PROTOCOL_ACK,