Re: Homegrown SIP load testing platform

[Joshua Goldbard <j () 2600hz com>]

Hey Jon,

This comes up on the voice ops list pretty regularly. Some folks have mentioned SIPVicious as a method for sip testing, 
but I think that's more for pentesting.

The Empirix stuff seems to be the state of the art today. On a previous thread I talked a bit about quality monitoring 
and why the stuff in the industry today isn't really giving you the kinds of feedback you're looking for, but load 
testing is a different problem.

If you do end up playing with the interrupt timers on the NICs, and you're successful, I'd love to hear what worked.

Some food for thought: we've got a set of tickets open with the TAC because a large router (sorry I don't have the 
model number) bricked in a repeatable fashion at 300 calls per second. It shouldn't be true, but sometimes the gateway 
device is the limitation, although I don't know if this is applicable in your example.

Anyways, I'm sorry I can't be of more help, but I personally see load testing at scale as a big unsolved problem for 
operators.

Cheers,
Joshua 

Sent from my iPad

On Jul 25, 2013, at 7:32 AM, "Jon Chleboun" <jon.chleboun () gmail com> wrote:

> I am interested to see if y'all have recommendations for putting together a
> SIP load testing platform using general purpose hardware and open-source
> (or inexpensive) software. We are aware of Empirix Hammer and similar
> solutions, and we are looking to see if there is an alternative option.
>
> Goals:
> - Generate somewhere on the order of 20k phone calls with real SIP and RTP.
> - Route the flows through our VoIP infrastructure to test performance
> limits.
> - Receive and analyze the SIP and RTP on the other end to find out at what
> load the signaling and/or media start to break down.
>
> Attempted already:
> - SIPp spread across many servers. Here the limiting factor seemed to be
> the CPU load from the interrupts from each packet. The CPU on the  servers
> sending and receiving the phone calls got bogged down before the VoIP core.
> - We have dabbled with interrupt moderation in the NIC drivers, but this
> has not seemed to help very much.
>
> Looks interesting:
> - Has anyone had success using PF_RING with Direct NIC Access and libzero
> from the folks at ntop? Has anyone been able to use this with SIPp or some
> other SIP and RTP generator?
>
>
> Many thanks,
>
> Jon Chleboun