Hi all
An update on the performance measurements:
The measurements I have referred to earlier all involved an Asterisk as
the call generator. Somehow this setup leads to extensive rtp bandwidth
usage. Each channel used around 500 kbps. If a phone is entered into the
loop, this is reduced to the expected 64 kbps. I have not found any
reason for this, but it certainly fouls up the test, and I have changed
the test setup.
Further, and since the earlier tests, the network has been updated to a
Gbits network.
I have now made two new test:
1) Using WinSIP from Touchstone as a call generator.
2) Using the Asterisk as one component, and setting up a chain of calls
which goes forth and back from the Asterisk and the FS. All call are
started from a real phone, and after 100 loops, where the calls are
answered and sent on by the dial plan, the calls are terminated by an
tone (<action application="gentones" data="%(500000,0,400)"/>) in the
FS.
The two test show similar top-figures at similar loads.
The first test would be my preferable, but it is limited to 50 calls due
to the trial licence limitations. Using an external non-FS and
non-Asterisk device will eliminate some uncertainties, that's why it
would be preferred.
The other test has been done with 600, 400 and 200 channels (300, 200
and 100 calls), and the results of the top command are:
cpu sy ni id wa hi si total
* 600 10 30 0 33 0 2 25 100
FS600 22 33 0 30 0 0 15 100
0
* 400 7 18 0 67 0 1 7 100
FS400 14 17 0 62 0 0 7 100
0
* 200 3 10 0 84 1 0 2 100
FS200 7 8 0 82 1 0 2 100
The results do not show significant differences between the capacity
behaviour of the Asterisk (*) and the FS. The also show an expected
interrupt load (si) proportional to the square of the call load.
Still the FS does not really outperform the Asterisk - which I find
disappointing. Any comments are welcome.
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