On Wed, Apr 21, 2021 at 5:19 PM Julien Chavanton <[email protected]> wrote:
> > 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. > > Nice! What I would liker to do, in an undefined future :), is to use the pjsip machinery for streaming audio from and to a file, so it will have their tried and true rtcp, rtcp-xr, etc implementation. It will be easier and more precise to combine these statistics to obtain better accuracy. > 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. > How do you judge sipjs implementation of jitter measurements? > Let's remember to keep each other posted if we improve this further. > > Definitely!!! And thanks again for VoIP Patrol! -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18
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