The easiest solution would probably be increasing the rtp-packetsize to 30-40ms instead of the default of 20. However, you're increasing the latency then. I suppose one could automatically start doing things like this based on rtcp. Michel On Thu, Oct 16, 2014 at 2:39 PM, Linux Vince <linuxv@gmail.com> wrote:
I have installed freeswitch and testing with SiPP
FreeSwitch was easily able to handle 1700 concurrent calls.
However when it starts going near 2000 my RTP packets are being lost and audio is totally gone in some cases.
How do i troubleshoot this ? And how to tune RTP performance for FreeSwitch.
I checked that CPU is not fully utilized and is somewhere around 40% range and memory is only at 10%.
Also network interface is not fully utilized as well as i was able to transfer files from server and interface was able to handle much more bandwidth than it was running.
Please help.
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