SIP Statistics Excellent, Good, Fair, Poor, Bad

Dear,
All Experts, I want to get each sip channel call quality, like Excellent, Good, Fair, Poor, Bad.

after I run this in Asterisk CLI “sip show channelstats” I get this:

Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
192.168.1.100 5a4e1c85211 00:15:48 0000047436 0000000000 ( 0.00%) 0.0000 0000000244 0000000000 ( 0.00%) 0.0008

now anyone can tell me how to determine Excellent, Good, Fair, Poor, Bad. I mean what is the percentage of determining Excellent, Good, Fair, Poor, Bad.

Thanks for Viewing Waiting for your replay…

The packet loss rates cannot be improved upon.

It is up to you to decide how much jitter is acceptable. I don’t think there is any official formula.

0 to 4 decimal places seems suspiciously good. Presumably 8ms is unlikely to cause any problems.

Thanks for your replay David

I have read this article https://www.voip-info.org/wiki/view/Asterisk+RTCP, where they talk about acceptable Packet Loss but not about JItter Loss.

Also here in this article https://kb.smartvox.co.uk/voip-sip/rtp-jitter-audio-quality-voip/ they talking about acceptable Packet Loss but not about JItter Loss.

Can you please tell me how can I determine the Live Call Quality, if there is anyway.

Listen to the audio!

Jitter is not a loss, but excessive jitter can cause losses. Jitter means that frames of speech samples aren’t arriving at regular intervals (normally 20ms). The receiver needs to delay the data sufficiently to allow for slowest arriving frame. If it gets the value too short, it can lose frames because it needs to use them before they have arrived. If it gets it too long, the long round trip time can be annoying and make residual echoes sound worse.

Also, if the jitter is very large, it may not have enough space to store very fast frames until the time to play them out arrives.

Thanks, ones again for your reply, please see below here is some data I am getting from my asterisk

Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
192.168.1.111 Oe6.cxcnFVg 00:05:03 0000014144 0000000977 ( 6.46%) 0.0000 0000000339 0000000078 (23.01%) 0.0143
192.168.1.100 fbdf6b6bbbc 00:02:52 0000008615 0000000000 ( 0.00%) 0.0000 0000000339 0000000000 ( 0.00%) 0.0007
2 active SIP channels

as you can see there is 2 SIP call one is with no loss at all & another one is with a loss, please let me know what is this 2 loss

And also look into this

Recv: Pack Lost ( %) 1st-Jitter Send: Pack Lost ( %) 2nd-Jitter
0000023013 0000000978 ( 4.08%) 0.0000 0000000339 0000000078 (23.01%) 0.0146

here is this 2 Jitter what is the work of this 2 jitter & effect on call-quality.