I have read this article https://www.voip-info.org/wiki/view/Asterisk+RTCP1, 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.
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.