innovailable / rtc-lib Goto Github PK

Hi,

I am trying to swap video streams in the middle of the call, because I need to change resolution dynamically. I found out, that probably only working solution is to remove old Stream from LocalPeer, add new Stream and then renegotiate with all RemotePeers.

Problem is, if I renegotiate only once after I removed old Stream and replaced it with new one, nothing happens.. RemotePeer thinks it is still the same Stream (with same stream.stream.id).

If I renegotiate twice: first time after removing old Stream and second time after adding new one, it could solve the problem, but another problem appears at the side of the peer which changed the stream.

After recieving an answer from first renegotiation:
Uncaught (in promise) DOMException: Failed to execute 'setLocalDescription' on 'RTCPeerConnection': Failed to set local offer sdp: The m= section with mid='1' should be rejected. from peer_connection.ts:342

After receiving an answer from second renegotiation:
Uncaught (in promise) DOMException: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Session error code: ERROR_CONTENT. Session error description: The m= section with mid='1' should be rejected.. from peer_connection.ts:293

It seems like a bug to me, or am I misunderstanding something?

This is problematic part of code:

 updateVideo(isHd) {
    let options = isHd ? this.hdVideoOptions() : this.basicVideoOptions();
    this.model.room.local.stream().then(s => {
        s.stop();
    });

    this.model.room.local.removeStream();
    this.renegotiateAllPeers(); // Renegotiation #1

    this.model.room.local.addStream(options)
    .then(newStream => {
        this.localStream = newStream;
        this.displayStream(this.localStream, $('#myVideo'));
        this.renegotiateAllPeers();  // Renegotiation #2
    })
    .catch((err) => console.error(err.message));
}

renegotiateAllPeers() {
    for (let peer of this.peers) {
        peer.negotiate();
    }
}

It is impossible to send message through DataChannel that I added to LocalPeer (Room.local.addDataChannel), because I don't have option to get this channel, when I want to call send() method on it.

Hello, I have some problems in doing a connection to an existing audio room. I'm creating an audio room, where two or more user can talk to each other:

This is my initial code on opening the page:

`
async initAudioRoom() {
const roomName = "raumname"

            const channel = new rtc.WebSocketChannel("wss://easy.innovailable.eu/" + encodeURI(roomName));
            const signaling = new rtc.MucSignaling(channel);
            const options = {
                stun: "stun:stun.innovailable.eu",
            }
            var room = new rtc.Room(signaling, options)
            this.room = room

            var current_this = this

            this.room.on('peer_joined', function(peer) {
                // create a video tag for the peer
                const view = $('<video playsinline autoplay>');
                $('#peers').append(view);
                const ve = new rtc.MediaDomElement(view[0], peer);
                // remove the tag after peer leftd
                peer.on('left', function() {
                    view.remove();
                });
            });
        },`

When someone wants to join the audio room I'm doing this:

`
const roomName = "raumname"

 try {
                // create a local stream from the users camera
                const stream = await this.room.local.addStream({ video: false, audio: true });

                // display that stream
                const ve = new rtc.MediaDomElement($('video'), stream);
            } catch(err) {
                alert("Unable to get stream, please give permission. You might have to go into settings if you denied before.");
                return;
            }

           try {
                await this.room.connect();
            } catch(err) {
                alert("Unable to join room: " + err.message)
            }

`

This works fine.. two or more user can talk to each other. Now I just try to define a function for a user, that only can listen to the users who are in the audio room.. how can I do this? The listening user have already the video tags of the peers, which are in the audio room, at the DOM. But the listening user can't hear anything. I tried something like a Listening Button and this function:

try { await this.room.join(); } catch(err) { alert("Unable to join room: " + err.message) }

Do you have an idea and could help me? Im trying it since days and im not able to finish that.

Thank you very much!

running make all outputs

node_modules/.bin/browserify --extension .ts -p tsify -t [ babelify --extensions .ts ] -s rtc -d --no-bundle-external src/index.ts -o dist/bundle/rtc.js
TypeScript error: /home/gsalvador/rtc-lib-master/node_modules/@types/browserify/index.d.ts(9,23): Error TS2688: Cannot find type definition file for 'node'.
make: *** [Makefile:49: dist/bundle/rtc.js] Error 1

I am experiencing weird bug, with rtc-lib using your public servers. After I connect in my web application to any room at your public WSS wss://easy.innovailable.eu and STUN stun:stun.innovailable.eu server. Everything works fine for 5 minutes and then rtc.Room instance fires closed event without detail. If any other peers are connected all our channels are closed as well.

I have tested it with my friends (Windows and Linux users) and no one has experienced the same bug. All of them were able to stay connected for whole session (up to 1 hour). I have tested it using Chrome and room closed always after exactly 300 seconds (5 minutes). When I tested it using Firefox room closed after 305 - 350 seconds.

When I monitor room WSS request on Network Developer Tools of Chrome it says "CAUTION: Request is not finished yet!" for 300 seconds (while connection works) and when caution disappears room closes. I don't understand why the request is not finished. Can't it be at your WSS side?

Do you have any idea what could cause this problem? 300 seconds doesn't seems like an coincidence, probably it is some timeout, but I have no idea where could be problem.

My setup:
OS: macOs Big Sur 11.1. (20C69)
Tested Browsers:
Google Chrome 88.0.4324.182 (Official Build) (x86_64)
Mozilla Firefox 86.0 (64-bit)

EDIT:
I resolved problem by pinging the signaling server.