[asterisk-users] [OT] Load testing with SIPp

[Olivier]

Hello,

I'm running load testing sessions.
My System Under Test is an asterisk 13 with 16GB, configured with maxfiles
set to 400 000.
This system is supposed do produce simple SIP trunking services without
transcoding.


The box sending call to my System Under Test is anabled with SIPp.
I'm banging on a 700 concurrent calls/50 CAPS limit I would like to
improve, if possible.

Tests are done with both signaling and media like this:

SIPp <---> SUT (asterisk 13) <---> Asterisk box echoing media

I checked bandwidth first and got 930 Mb/s on each leg (from SIPp to SUT or
SUT to echoing box) using iperf3 TCP testing though my target relies on UDP


My questions are:

1. Have you ever noticed a better scalability using UDP or TCP ?

2. Where do Retransmission I'm observing on SIPp console most probably come
from ? Network issues ? My SIPp not beeing correctly tuned ? Lack of
resources somewhere ?

3. Recommandations ? Suggestions ?

Best
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180306/fd47beaf/attachment.html>