I recently changed something in the WebsocketServer implementation of one of my projects, that (presumably) decreases the Round-Trip-Time (RTT) of its messages. I want to measure the RTT of both implementations to compare them. For this, I am sending m messages of s bytes from c different connections.
The thing I am wondering now is: Should the RTT be measured for each message separately - remembering the sending time of each message and when the response arrives -, or should I remember the time the first message was sent and the response to the last message arrives? Which one is more accurate?
I would probably go for the first approach, what made me wondering is that websocket-benchmark seems to use the latter approach. Is this just an oversight or is there a reasoning behind it?