Ant-Media-Server WebRTC playback stops after some hours after upgrading the server.

One of the users has an issue where the stream stops to play after 12 to 24 hours.
The publish and play protocol used is WebRTC.
The stream doesn't play on Origin as well as Edge.
The issue happens after upgrading the server to version 2.8.2.
When rolled back to the previous version 2.6.x, the issue is not happening.

Sample logs 2024-04-16 18:26:56,911 [DecodingQueue - 383684] INFO i.a.e.adaptive.WebRTCVideoForwarder - Dropping video frame for stream:ctc_bugmv. Video/audio enabled:true/true videoTimeMS: 55191303 audio duration:-61986493 total frame drop count:132020 dropVideoFrameUntilKeyFrame:true 2024-04-16 18:26:56,949 [DecodingQueue - 383105] INFO i.a.e.adaptive.WebRTCVideoForwarder - Dropping video frame for stream:ctc_bitfiremv1. Video/audio enabled:true/true videoTimeMS: 23353903 audio duration:-66124786 total frame drop count:42260 dropVideoFrameUntilKeyFrame:true 2024-04-16 18:26:57,245 [DecodingQueue - 383684] INFO i.a.e.adaptive.WebRTCVideoForwarder - Dropping video frame for stream:ctc_bugmv. Video/audio enabled:true/true videoTimeMS: 55191636 audio duration:-61986173 total frame drop count:132040 dropVideoFrameUntilKeyFrame:true 2024-04-16 18:26:57,281 [DecodingQueue - 383105] INFO i.a.e.adaptive.WebRTCVideoForwarder - Dropping video frame for stream:ctc_bitfiremv1. Video/audio enabled:true/true videoTimeMS: 23354236 audio duration:-66124466 total frame drop count:42280 dropVideoFrameUntilKeyFrame:true

We have access to the environment.

Apr 23 '24 09:04 Mohit-3196

Possibly also relevant:

All WebRTC traffic routes through a TURN server

Audio bitrate is raised up from the default upon publish: `var audioBitrate = 256000;

function setAudioBitrate() {
    var state = webRTCAdaptor.signallingState(streamId);
        if (state != null && state != "closed") {
            var iceState = webRTCAdaptor.iceConnectionState(streamId);
            if (iceState != null && iceState != "failed" && iceState != "disconnected") {
                var params = webRTCAdaptor.remotePeerConnection[streamId].getSenders()[0].getParameters();
                params.encodings[0].maxBitrate=audioBitrate;
                webRTCAdaptor.remotePeerConnection[streamId].getSenders()[0].setParameters(params);
            }
        }
    console.warn("Changing outbound audio bitrate to "+(audioBitrate/1000)+"kbps");
}

`

Everything else is pretty standard from the default configuration.