OME WebRTC stream drifting away from realtime

[devedse]

Describe the bug
I'm currently using OpenStreamingPlatform to stream games together with my friends. However over the span of about 30-minutes+ the stream starts being less and less realtime. Initially there's about 0.1-0.2 seconds delay but later this becomes 4 seconds or more.

If you refresh the page however, the delay is instantly gone.

To Reproduce
Steps to reproduce the behavior:

• All my settings are written down in this Discussion: OME worked before, but can't get it working behind double NAT with Tunnel and VPN to route traffic #1238

Expected behavior
I would expect the stream to stay live as long as I'm watching

Logs
See: #1238

Server (please complete the following information):

• OS: Raspberry PI 4 ARM 64 + docker
• OvenMediaEngine Version: 0.15.12
• Branch: N/A

Player (please complete the following information):

• Device: Windows PC
• OS: Windows 11
• Browser: Chrome
• Version: Latest

Additional context
N/A

[getroot]

This is most likely not an issue with OvenMediaEngine. Or do you have reason to believe this is a bug or issue in OME? If so, please share that information.

It could be if the network speed between OME and Player is not fast enough and you are playing with WebRTC/TCP. (because TCP retransmits)

Try lowering the bitrates of the output and playing it. Or try playing with WebRTC/UDP.

[devedse]

I started a new stream with baseline set to main:

0.35 delay initially: (I didn't reset the timer)

0.45 about 6 minutes later:

0.54 about 30 minutes later:

[getroot]

Your browser's jitter buffer has increased, and 0.54 seconds seems to be an acceptable latency that can happen depending on network jitter. Looking at the graph, it seems that your jitterbuffer size converges around that. For reference, Chrome's jitter buffer is a value that Chrome monitors various conditions and dynamically adjusts internally, so it cannot be controlled externally.

After more time (how much?) in this state, does the latency continue to increase above 4 seconds?

[devedse]

I tried it yesterday with a friend of mine for about an hour but we couldn't get the latency significantly higher. I'll do some more research into this and will come back to you with new results over the next few days.

[getroot]

Yes, I hope the information below will help you.

• If you play with WebRTC/TCP on a device with a slow network, you won't experience any stuttering, but the latency will continue to increase.

• If you play with WebRTC/UDP on a device with a slow network, the latency won't increase but you will experience stuttering.

[devedse]

That makes sense, I'll keep you posted :).

Wouldn't it be possible though to increase the playspeed of WebRTC/TCP when the buffer get's full to be able to catch up? Or is that logic also hidden in Chromium?

[getroot]

And you wrote OME RTSP in the title of this issue, is OME WebRTC correct? If so, please edit the title.

[devedse]

You're fully correct, I've modified the title