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

[Bruce Ferrell]

On 03/06/2018 01:58 PM, Olivier wrote:
> 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
>
>
I do network management for a living.

In your description, I see nothing to describe the network other than an observed 930Mb/s.

What is the network configuration;  What NIC(s), switches etc.

Treating these as effectively "unlimited" is a certain recipe for banging into unexpected limits.

Different NICs and switchs can and do provide differing levels of performance.