Re: [SR-Users] [ANNOUNCE]: sipnagios, a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc)

Few notes on the mos-lq (listening quality), it consider both losses from jitter (discarded) and never received. I tried to keep the equation and variables as defined in the ITU, but it is relatively simple in the end. One thing missing is the delay impairment to have mos-cq this would be RTT + jitter buffer size of both endpoints. This way we will correctly account for jitter impairment in terms of loss and delay.

More context on jitter, as I recently went back looking at some MOS score computation. Since we compute MOS in the endpoint it can be more precise when it comes to jitter. In most cases, when done in a relay, I found that jitter is hard (or not accounted) for properly, since we extrapolate adaptive / static buffers that will receive the packets. What I found in most cases was jitter x 2 like in rtp-engine seems like the best option but should endup underestimating the impairment as this would mean an adaptive buffer and assume not too much jitter of jitter meaning the size of the buffer based on estimate is always fine with the given jitter and not dropping late packets and it must drop packets when it shrinks.

Let's remember to keep each other posted if we improve this further.