11 Mar
2004
11 Mar
'04
9:12 a.m.
On Mar 11, 2004 at 07:53, Alex Bligh <alex(a)alex.org.uk> wrote:
Before I reinvent the wheel and go write it, is there
a utility which will
simulate making & receiving calls (i.e. SIP conversations, **plus RTP**) in
scale (to answer a question like "will my SIP+RTPproxy deployment scale for
{100,1000,10000,100000} users using the following parameters for average
call duration, % phones in use etc."
No, at least not a free one. It will be really great to have one.
If not, anyone have any recommendations for a (free) stack that does a
decent UAC/UAS job, including audio, and will run without GUI?
Be carefull when you choose a stack. It must be fast, or the bottleneck
will be your testing tool. We had a few surprises with ser, which is
faster in general than the testing tools on the same hardware.
In the rtpproxy case the bottleneck would be the number of nathelper
commands it can process and not forwarding of the rtp traffic. I did a
simple test on a gigabit interface and rtpproxy was able to forward
60Mb/s on an Athlon 2000+. This was on a single session, having a lot of
simultaneous sessions will decrease its forwarding capability (lots of
time spent in pool), but I still don't think this will be the
bottleneck.
Andrei