SIP Load test

I am in the middle of programming a simple SIP load testing application where I generate x number of calls to a trixbox system and send the milliwatt 1k tone for t duration. Both x and t are currently manually changed in the application to match the needed load. The trixbox unit under test is currently recording all the calls so I can verify call quality by playing back the recordings.

Now the interesting part: If I set the call to last 2 minutes the subsequent recorded file has a duration for nearly 10 minutes??? The calls are confirmed to last the pre-configured duration via the CDR. Why is it transferring more data than it should? Or maybe I should ask How do I throttle the data flow to match a 64k voice stream? My assumption (also my first error is to assume), SIP and the codec would maintain the 64k rate to the recording.

Your input is greatly appreciated.

Try asking this question on the asterisk users list.