[VoiceOps] Homegrown SIP load testing platform

[Erik Flournoy]

Yate




On Tue, Jul 23, 2013 at 3:57 AM, Jon Chleboun <jchleboun at e-c-group.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
>
>
>
>
>
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