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Today's Topics:
1. Re: Edgewater competitors (My List Account)
2. Homegrown SIP load testing platform (Jon Chleboun)
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Message: 1
Date: Mon, 22 Jul 2013 11:47:50 -0400
From: "My List Account" <[email protected]>
To: "'VoiceOps'" <[email protected]>
Subject: Re: [VoiceOps] Edgewater competitors
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
+1 for the TA900s and their warranty. They really don't care where you got
them from (reseller or ebay) when it comes time to send one in for repair.
Unless it's very obvious lighting damage they will repair almost anything
that is sent in. I've yet to come across any type of service that can't be
converted to SIP with a TA900.
Richey
-----Original Message-----
From: VoiceOps [mailto:[email protected]] On Behalf Of Jay
Hennigan
Sent: Monday, July 22, 2013 12:47 AM
To: VoiceOps
Subject: Re: [VoiceOps] Edgewater competitors
On 7/21/13 4:06 PM, PE wrote:
> We've been using Edgemarcs for years and it's time to consider a
technology refresh. What's everyone using? Looking for pros, cons, and
opinions.
Adtran TA900 series. We've changed out just about all of our Edgemarcs for
them over the last several years.
Great tech support, 10-year hardware warranty, they just work.
Best of all, no "crippleware" where you are forced to pay extra for turning
on a port or exceeding some arbitrary number of calls.
--
Jay Hennigan - CCIE #7880 - Network Engineering - [email protected] Impulse
Internet Service - http://www.impulse.net/ Your local telephone and
internet company - 805 884-6323 - WB6RDV
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------------------------------
Message: 2
Date: Tue, 23 Jul 2013 09:57:50 -0400
From: Jon Chleboun <[email protected]>
To: [email protected]
Subject: [VoiceOps] Homegrown SIP load testing platform
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
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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End of VoiceOps Digest, Vol 49, Issue 2
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