[asterisk-dev] asterisk with osp performance test results

John Todd jtodd at digium.com
Thu Oct 23 00:47:41 CDT 2008

I would not expect NAT to create any significant decrease in  
performance on these results, since the packet re-write on the RTP  
packets is fairly lightweight from what I understand.


On Oct 22, 2008, at 5:12 PM, Gregory Boehnlein wrote:

> Hmm.. I would be very interested to see how introducing NAT into the  
> picture and having Asterisk do the actual NAT fixups for the media  
> stream would impact this (if at all).
> From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com 
> ] On Behalf Of Di-Shi Sun
> Sent: Wednesday, October 22, 2008 7:36 PM
> To: asterisk-dev at lists.digium.com
> Subject: [asterisk-dev] asterisk with osp performance test results
> All,
> In 2007 we published the results of a performance test of Asterisk  
> as a Back to Back User Agent (B2BUA). Based on the helpful feedback  
> from the mailing list we improved our test procedures and re-ran a  
> new performance test case of Asterisk configured as a B2BUA.
> Our test platform hosting Asterisk was a $1000 Dell PowerEdge 840  
> with a Quad Core Xeon X3220, 2x4M cache, 2.40 GHz, 1066 MHz FSB and  
> 4 GB RAM. Redhat V5 was the operating system. The test was  
> configured to simulate a wholesale VoIP operation with three minute  
> call durations and an average of two call retries for every  
> completed call. This was an "out of the box" Asterisk configuration  
> with default settings and no optimizations.
> We found that Asterisk on the test server could handle approximately  
> 1000 simultaneous calls with no codec transalation. This works out  
> to be about a $1 per port investment for a B2BUA platform.
> When calls were transcoded from G.711 to G.729, the call capacity  
> fell to 320 simultaneous calls. With the added cost of of the G.729  
> codec royalty and the lower call capacity, the cost increases to  
> approximately $13.50 per port
> You can download the test results and all the test plan details from:
> http://www.transnexus.com/White%20Papers/Performance_Test_of_Asterisk_v1-4.htm
> Di-Shi Sun
> VoIP Routing, Accounting, Security
> www.TransNexus.com

John Todd
jtodd at digium.com        +1-256-428-6083
Asterisk Open Source Community Director

More information about the asterisk-dev mailing list