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I am using CNLab's UX desktop app to test my internet connection's consistency as shown below for the upload. What is the difference between server-side and client-side data transfer rates. For VOIP/videoconferencing applications, which measure is more important to be consistent in an internet connection? With regard to my test setup, I am connected via WLAN.
"Client" or "server" are concepts that don't exist on the network that you might be watching. What matters is the direction of (main) data flow, so "sender" and "receiver" might be more useful.
Generally, the sender sees more traffic. The receiver sees the sender's traffic minus the lost/dropped packets.
Ideally, the difference is low but that depends on the situation and the used protocols. Which side matters more to you cannot be universally answered - you might need to analyze both, especially for an application like VoIP. VoIP is most sensitive to latency and jitter, less so to packet loss (within a reasonable margin). Note that applications and protocols above the transport layer are explicitly off-topic here, however.
In your diagram, the sender (server) seems to transmit a constant data flow. The receiver (client) sees fluctuating rates - most likely due to variable queueing and congestion along the path (assuming there's no retransmission).
You're testing with TCP as transport protocol which may not be meaningful for VoIP.
