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sipXecs SIP Trunking Performance Looking Good PDF Print
Written by Administrator   
Wednesday, 22 April 2009 00:43

We did some testing of the new sipXecs IP PBX SIP trunking gateway and thought you might be interested in the results. It's looking really good. Initially we were shooting for a guaranteed sustained rate of 200 simultaneous trunks. The test we performed loaded the system with 450 simultaneous trunk calls (~20 T1 trunks). At that rate it consumed 16.25% total CPU (8.25% application load) and a total of only 47 MB of memory. This means that there is plenty of CPU capacity and memory left to run all the other sipXecs components in parallel with SIP trunking.

The table below shows the detailed test results. The tracker issue XECS-1577 provides the entire report.

sipXecs SIP trunking performance analysis

The test system was a Dell Optiplex 755 with Intel Core2 Duo E6550 CPU @ 2.33 GHz, 1333 MHz FSB and 2 GB RAM @ 677 MHz. sipXecs build 3.11.9-014397 was used. This is a very cheap small form factor (SFF) Dell desktop PC.

The actual component that causes CPU load in this test is the media anchoring relay. A media relay is required so that the SIP trunking gateway can perform the necessary NAT compensation. In a high-availability (HA) configuration using two sipXecs call control servers, there are also two media relay components. This not only doubles the available capacity for SIP trunking, but also provides redundancy. In addition, the media relay component is capable of doing both near-end and far-end NAT detection and compensation, enabling seamless remote worker support. While we have not pushed the test to its limit in this setup it would be interesting to see whether audio quality can hold up at a significantly higher number of trunks. What you can see in the above report is that the packet arrival jitter only goes up a little bit as we increased call volume, rendering toll quality voice throughout the test.

The test was done using the G.711 codec. The SIP trunking gateway would also process wide-band audio (HD Voice) using G.722 or any other codec. While this is typically not supported by the ITSP, it could be used for site-to-site trunks if there are NATs involved that require a media relay.

The list of currently certified ITSPs is work in progress. You can participate in the interop program if you'd like your favorite provider certified.

Try it out and report back.

 
 

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