[Freeswitch-users] Conference RTP issue

[Sergey Safarov]

Hi all
I have broken audio for conference call for first 4-7 seconds of call.
That is paging call using FS conference.
Caller make call to conference and then FS conference dials about 28 phones
with "Alert-Info: <intercom>" header (auto answer for Yealink phones).

If you look into RTP stats using Wireshark, then you can see high jitter
value for RTP stream from  FreeSwitch. Oh jitter graph value is variable.

Is anybody observe similar issue?
How i can troubleshoot high jitter issue on FS side?

call example is attached
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