On Mon, 2004-05-10 at 16:16, John Todd wrote:
> http://sipp.sourceforge.net/
>
> Anyone care to throw this at Asterisk to see what happens? I would,
> but I am having significant temporal shortfalls recently due to the
> apparent warping of the space/time continuum when I answer the phone
> with clients/associates. It seems that entire days pass by before I
> hang up... very odd, and very counter-productive to getting good
> Asterisk work done.
Ok. Test report:
I set up an UAC which was generating 10cps of 10s duration and the
corresponding UAS which received this calls. The command used to
generate the calls which were GSM was:
sipp 192.168.65.100 -s 700 -sf uac.xml -d 10000 -r 10
The command to receive the calls on another box was:
sipp -sf uas.xml
IÂm using my own uac.xml and uas.xml just to talk GSM, I monitored
using my 7960 agains a MusicOnHold.
On my Xeon 2.4Ghz no call were dropped and no audio problems. Note that
I use nat=yes and canreinvite=no for UAC/UAS on sip.conf. It seems that
SIPP doesnÂt support authentication for now.
For 40cps of 10sec duration (which means 400 concurrent calls) it works
just fine for me. At 50cps of 10sec duration no call are dropped but I
start seeing some SIP packets retransmitions.
At 60cps lots of call gets dropped but the funny thing is that the
audio through the 7960 isnÂt much affected.
Real nice tool.
--
Juanjo sin .sig
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