Packet loss with local WebRTC clients

Reedz

8/15/24, 7:47 AM

I stream live audio to clients via WebRTC, and a streaming app called Barix on a local network for translated events.

When there are more than 20 or so clients, the network undergoes significant packet loss at a start of the event, for the first 10-15mn, resulting in the audio cutting and a lot of packet loss if I try to ping any resource on the network.

There are periods of packet loss at other times without any apparent reason as well. The network is disconnected from the internet, and the only resource available on the network is the audio streaming, which doesn't theoretically use much bandwidth. Clients connect via gigabit APs to an Edgerouter X.

Do you know what could cause this or how I could effectively diagnose the issue?

0 + 2

networking

packetloss

webrtc

Jaromanda X

8/15/24, 8:31 AM

so clients are wireless (i.e. Wi-Fi)? Perhaps someone uses a microwave :p What bitrate is the audio stream? Is the source connected through wifi as well?

Reedz

8/15/24, 10:15 AM

All clients are connecting with Wi-Fi. I am streaming with ffmpeg using the s16le format, I am not sure what the bitrate is as I kept the default ffmpeg command but I don't think it may be the issue as the traffic doesn't exceed 1mb/s on my EdgeRouter when all clients are connected. Still, I don't know what happens to the traffic during the bad episodes as I can't access the router interface anymore at that point.

Elon Musk

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