I see tremendous value in load testing before installation. Regarding SIP calls, does anyone have a good set of tools or favorite utility for:
If it can provide statistic too (packet loss, latency, jitter, MOS, etc.), even better, although, we could see these stats in AOS' superb VQM. In theory, we should be able to see how many concurrent calls a network path or segment will support (or whether a given path will support an intended planned amount of call volume).
Thanks in advance for any ideas! Free is always great, but we don't mind paying for something that works well.
First, thanks for posting in our Support Community! ADTRAN does not manufacture any hardware or write any software to do bulk call testing. However, all of our units are fully tested against an Abacus bulk call generator. My guess is that this system is more that you need, but you could check into it. SIPp is another option you might consider, but I have only used it to generate specific SIP packets; I've never tried its RTP functionality.
Another idea would be to put two Adtran units in a back-to-back configuration and setup calls with either loopback accounts or voice users. Each unit would be setup with a SIP server defined as the other unit's IP address. Note that loopback accounts are limited to five users/calls per unit. This option would take more time to setup, but is definitely feasible. Once the calls were setup, you could then check our Voice Quality Monitoring feature (VQM) for MOS scores, delay, jitter, etc.
i'm agree with @david Spirent Abacus is excellent for do Stress Tests, particularly the Abacus 5000 can allow you to test either E1 links or Ethernet VoIP like SIP.
I am trying to get my head around SIPp, however, it's a bit technical so far. Thanks for the suggestions. I'll check into those products!
David, I'm very intrigued about setting up put two ADTRAN units in a back-to-back configuration in order to setup calls with loopback accounts or voice users. I don't think I've seen this before. I have a JDSU T-BERD 5800 that can simulate a PBX and I've been trying to find a way to pass SIP traffic with an ADTRAN(s) TA 900 series or 3430 in an AD HOC Lab without a soft switch (NO ISP) I haven't been very successful. Is there any documentation that would be able to assist with what you're describing?