Rtp jitter buffer

Hello every one…
how we can monitor RTP live status like jitter buffer latency ms or packet loss voice qulifty excellent, good, or poor on any specific channel … ?

rtcp set debug ip

rnun calls, and you will see info in cli :


  • Sent RTCP SR to
    Our SSRC: 1169877703
    Sent(NTP): 1600379656.791070
    Sent(RTP): 40000
    Sent packets: 250
    Sent octets: 40000
    Report block:
    Their SSRC: 939789714
    Fraction lost: 0
    Cumulative loss: 0
    Highest seq no: 61194
    IA jitter: 0.0016
    Their last SR: 0
    DLSR: 0.0000 (sec)

more info you can find: https://www.voip-info.org/asterisk-rtcp/

Jitter buffers are not used in default configurations where RTP is used on both sides. The destination VoIP device iis expected to provide the jitter buffering.

I’m not sure if they are forced for conference bridging.

thanks guys… for replay .

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