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Initial Support for Spacebar WebRTC #1108

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@CE1CECL CE1CECL commented Dec 28, 2023

please note: WebRTC was ONLY tested using a test client that was since deleted by spacebar owners, no other clients were tested, nor bot libraries were either (discord.js didn't work for me since it always requires a https connection)

NOTE: This PR DOES NOT add Screen sharing, but adds webcam support and voice
BREAKING CHANGES: Windows can no longer be supported after this PR is merged, because medooze-media-server doesn't support windows WebRTC can now be disabled with a env var to continue support for Windows

Known bugs:

  • Muting, unmuting, deafening, and undeafening makes the voice call reconnect for unknown reason.
  • Defend does not work on test client because of the reconnect bug.
  • Video support can be buggy, requiring both cameras on to fully work (why?)
  • You tell me what other issues there are: this has been a worked on thing by Me, @MaddyUnderStars & was initially created by @SamuelScheit

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Needs a tiny amount of work :)

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@Puyodead1
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if there isn't already, there should be a way to disable webrtc from loading, because I use windows for development, and I am not switching anytime soon. either config, or at the very least an environment variable

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Worst case you could use WSL I guess?
But yeah, this is probably important, even if Windows remains a minimally supported OS.

Signed-off-by: Christopher Lentocha <[email protected]>
@CE1CECL CE1CECL changed the title Initial Support for Spacebar WebRTC & Dynamic URL Support Initial Support for Spacebar WebRTC Dec 28, 2023
Signed-off-by: Christopher Lentocha <[email protected]>
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CE1CECL commented Dec 29, 2023

if there isn't already, there should be a way to disable webrtc from loading, because I use windows for development, and I am not switching anytime soon. either config, or at the very least an environment variable

Added but it needs testing directly on Windows, which I don't have, I have Linux.

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CE1CECL commented Dec 29, 2023

Worst case you could use WSL I guess? But yeah, this is probably important, even if Windows remains a minimally supported OS.

No. WSL Is NOT a option, WSL requires Hyper-v which is garbage in general,, slow, etc... WSL is more like emulation than a layer like WineHQ

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what? Wine is a stdlib overlay + bincompat layer, WSL is a full on vm (and it isnt slow either)

@CE1CECL CE1CECL marked this pull request as ready for review December 30, 2023 21:38
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CE1CECL commented Dec 30, 2023

Marked as ready since its possible to disable webrtc

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You don't need to include a separate file for windows, you could use dynamic imports or just, a top level if statement?

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You don't need to include a separate file for windows, you could use dynamic imports or just, a top level if statement?

I tried, it doesn't seem to work out, because import is required to be used (const require doesn't seem to work) and needs to be at the top without an "if" statement. What I did in my last commit was the easiest and probably cleanest, and understandable by someone else.

You can just (if you want)
diff -aprNu src/bundle/Server.ts src/bundle/WinServer.ts

guild_id: voiceState.guild_id,
user_id: this.user_id,
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I believe this is incorrect. That parameter is used to determine the recipients of this gateway event. Your edit only sends the event to the user who left the channel, which means other users in the voice channel won't be updated.

@@ -84,7 +84,7 @@ export class VoiceState extends BaseClass {
@Column({ nullable: true })
self_stream?: boolean;

@Column()
@Column({ nullable: true })
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This doesn't need to be nullable, it should just be set to false by default. Regardless, you didn't write migrations for this change.

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@CE1CECL CE1CECL Jan 5, 2024

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This doesn't need to be nullable, it should just be set to false by default.

What do you mean by "false by default"?

Regardless, you didn't write migrations for this change.

Wasn't sure if I would keep the change but I will write mig's once the first question gets answered.

Comment on lines -62 to +63
server: this.server,
noServer: true,
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Rory&, this was done so that the websocket server would not be responsible for upgrading http connections, which is now done above with the uncommented update listener. This was done so that voice ws is mounted on /voice

I don't think this is necessary though, I would rather it just use a different port? This makes routing through nginx less difficult when using bundle.

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Worst case you could use WSL I guess? But yeah, this is probably important, even if Windows remains a minimally supported OS.

No. WSL Is NOT a option, WSL requires Hyper-v which is garbage in general,, slow, etc... WSL is more like emulation than a layer like WineHQ

WSL 2 needs Hyper-v, the original WSL is to windows what wine is to linux

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honestly, hyper-v/WSL2 is completely fine, the slow part is the 9PFS mount for your windows drive

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CE1CECL commented Nov 22, 2024

Worst case you could use WSL I guess? But yeah, this is probably important, even if Windows remains a minimally supported OS.

No. WSL Is NOT a option, WSL requires Hyper-v which is garbage in general,, slow, etc... WSL is more like emulation than a layer like WineHQ

WSL 2 needs Hyper-v, the original WSL is to windows what wine is to linux

I wonder if the original WSL will work or not with this? It should, and if it does, then users should use that to enable webrtc until a real native fix comes out.

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this hasn't had work for quite a while, there's a different WebRTC branch now that has a different blocker, and most of us don't know enough rust to fix it

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CE1CECL commented Nov 22, 2024

this hasn't had work for quite a while, there's a different WebRTC branch now that has a different blocker, and most of us don't know enough rust to fix it

Ah, I see! Is there a reason this was re-done from scratch?

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not 100% sure tbh, I just want to see WebRTC working lol

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CE1CECL commented Nov 22, 2024

not 100% sure tbh, I just want to see WebRTC working lol

May I ask what on this PR isnt working exactly?

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not sure what is wrong with it, I'm not the person who deals with this stuff, I just make a client lol

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dank074 commented Nov 23, 2024

I just tried this now and it seems to work just fine on WSL at least for testing purposes for those who develop on Windows. Luckily VS Code makes it easy to develop on WSL. I don't know if anyone was ever planning to use Windows for production anyway.

Issues I found:

  • Like the PR description states, both cameras have be turned on, but even then, it shows you your own camera instead of the other user's camera. I think it might be doing the same for sound, it's just harder to test sound.
    EDIT: It's a bug when calling attachTrack you are attaching it to the current client and not the other clients in the call. I changed it and now video is working as expected, even without having both cameras on

  • Reconnect bug was solved by puyo in his branch dev/webrtc (his changes on VoiceStateUpdate.ts seemed to solve the issue for this branch as well)
    changes.patch

  • H264 seems to work for me. I just had to modify the SDP.json file and match Discord parameters

  • Codec id is hardcoded in SDP.json but it should be dynamic since it will differ between Firefox and Chrome. SelectProtocol message should determine the codec id

  • Cannot build Spacebar on Windows since medooze cannot be installed on Windows at all (node-gyp build fails)

I think a possible solution for that last bullet point is probably to make the webrtc server its own separate package which runs independently and communicates with the spacebar instance using RabbitMQ. This will mean that Spacebar will be webrtc-server implementation agnostic, and the end user can choose which implementation to pair with Spacebar. It will also mean that the webrtc server won't necessarily have to be a NodeJS server.

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CE1CECL commented Nov 24, 2024

I just tried this now and it seems to work just fine on WSL at least for testing purposes for those who develop on Windows. Luckily VS Code makes it easy to develop on WSL. I don't know if anyone was ever planning to use Windows for production anyway.

Issues I found:

* Like the PR description states, both cameras have be turned on, but even then, it shows you your own camera instead of the other user's camera. I think it might be doing the same for sound, it's just harder to test sound.
  EDIT: It's a bug when calling `attachTrack` you are attaching it to the current client and not the other clients in the call. I changed it and now video is working as expected, even without having both cameras on

May I ask what changes were exactly done?

* Reconnect bug was solved by puyo in his branch dev/webrtc (his changes on VoiceStateUpdate.ts seemed to solve the issue for this branch as well)
  [changes.patch](https://github.com/user-attachments/files/17882959/changes.patch)

I saw this is the same changes as the portion from 6beee77 thanks! I'll add it to the branch after I test it and fix the merge conflicts.

* H264 seems to work for me. I just had to modify the SDP.json file and match Discord parameters

* Codec id is hardcoded in SDP.json but it should be dynamic since it will differ between Firefox and Chrome. SelectProtocol message should determine the codec id

* Cannot build Spacebar on Windows since medooze cannot be installed on Windows at all (node-gyp build fails)

I think a possible solution for that last bullet point is probably to make the webrtc server its own separate package which runs independently and communicates with the spacebar instance using RabbitMQ. This will mean that Spacebar will be webrtc-server implementation agnostic, and the end user can choose which implementation to pair with Spacebar. It will also mean that the webrtc server won't necessarily have to be a NodeJS server.

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CE1CECL commented Nov 24, 2024

Unless you did this, this is my current guess:

diff --git a/src/webrtc/opcodes/Video.ts b/src/webrtc/opcodes/Video.ts
index aa97c18..c451063 100644
--- a/src/webrtc/opcodes/Video.ts
+++ b/src/webrtc/opcodes/Video.ts
@@ -76,7 +76,7 @@ function createStream(
                if (!client.in.stream) return;
 
                client.in.stream?.getTracks().forEach((track) => {
-                       attachTrack.call(this, track, client.websocket.user_id);
+                       attachTrack.call(this, track, this.user_id);
                });
        });
 }
@@ -165,7 +165,7 @@ function handleSSRC(this: WebSocket, type: "audio" | "video", ssrcs: SSRCs) {
                        if (client.websocket.user_id === this.user_id) return;
                        if (!client.out.stream) return;
 
-                       attachTrack.call(this, track, client.websocket.user_id);
+                       attachTrack.call(this, track, this.user_id);
                });
        }
 }

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dank074 commented Nov 24, 2024

Unless you did this, this is my current guess:

diff --git a/src/webrtc/opcodes/Video.ts b/src/webrtc/opcodes/Video.ts
index aa97c18..c451063 100644
--- a/src/webrtc/opcodes/Video.ts
+++ b/src/webrtc/opcodes/Video.ts
@@ -76,7 +76,7 @@ function createStream(
                if (!client.in.stream) return;
 
                client.in.stream?.getTracks().forEach((track) => {
-                       attachTrack.call(this, track, client.websocket.user_id);
+                       attachTrack.call(this, track, this.user_id);
                });
        });
 }
@@ -165,7 +165,7 @@ function handleSSRC(this: WebSocket, type: "audio" | "video", ssrcs: SSRCs) {
                        if (client.websocket.user_id === this.user_id) return;
                        if (!client.out.stream) return;
 
-                       attachTrack.call(this, track, client.websocket.user_id);
+                       attachTrack.call(this, track, this.user_id);
                });
        }
 }

I just did this

diff --git a/src/webrtc/opcodes/Video.ts b/src/webrtc/opcodes/Video.ts
index 1a288f8e..2d646d4b 100644
--- a/src/webrtc/opcodes/Video.ts
+++ b/src/webrtc/opcodes/Video.ts
@@ -165,7 +165,7 @@ function handleSSRC(this: WebSocket, type: "audio" | "video", ssrcs: SSRCs) {
 			if (client.websocket.user_id === this.user_id) return;
 			if (!client.out.stream) return;
 
-			attachTrack.call(this, track, client.websocket.user_id);
+			attachTrack.call(client.websocket, track, this.user_id);
 		});
 	}
 }

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