Synchronization problem on multi-stream mixing

[djuka]

I am using the 1843 version.
Is it possible to use the property:

enable_empty_shift_writer=true

since some multiple-stream recordings have synchronization problems between video and audio.

And, is there a possibility to get all mediaSessionIds of all participants in the room using Web SDK on front app?

 

[Max]

Good day/

enable_empty_shift_writer=true

This setting is not available since a very old builds. You cannot use it.

since some multiple-stream recordings have synchronization problems between video and audio.

This is probably due to bad channel quality of a source stream publishing. You can use lower resolution/bitrate to publish a stream or switch to TCP transport

ice_tcp_transport=true

 

[djuka]

I added this to flashphoner.properties but streams still goes over UDP

 

[Max]

And, is there a possibility to get all mediaSessionIds of all participants in the room using Web SDK on front app?

Media session id is the server side entity. You should use REST API /stream/find to find participant streams by name, and get media session id from response

POST /rest-api/stream/find HTTP/1.1
Host: 192.168.1.101:8081
Content-Type: application/json
 
{
    "name":"room-aaf982-user2-3df5",
    "published":true
}

 

[Max]

I added this to flashphoner.properties but streams still goes over UDP

Did you restart WCS? The option requires restart to be applied

 

[djuka]

Yes, I did.

"audioCodec": "opus/48000/2",
        "videoCodec": "H264",
        "record": true,
        "width": 640,
        "height": 480,
        "bitrate": 0,
        "minBitrate": 0,
        "maxBitrate": 0,
        "quality": 0,
        "history": false,
        "gop": 0,
        "fps": 0,
        "audioBitrate": 0,
        "codecImpl": "",
        "transport": "UDP",
        "cvoExtension": false,
        "createDate": 1709036797568,
        "mediaType": "publish",
        "audioState": {
            "muted": false
        },
        "videoState": {
            "muted": false
        },
        "mediaProvider": "WebRTC",

this is capture of one published stream from /stream/find_all

there is transport UDP still

 

[Max]

Please check if you set the transport explicitly at frontend with stream option:

    room.publish({
        display: display,
        constraints: constraints,
        record: false,
        receiveVideo: false,
        receiveAudio: false
        ...
        transport: "UDP",
        ...
    });

If yes, set it to TCP.