On 02/09/2012 01:17 PM, Danny Nicholas wrote:
If the MOH thing is really true, a more "realistic" test would be to run
playback(demo-instruct). Since I know that I will eventually cross this
bridge in real life/real time, I devised this test on my Asterisk 10.0 box
Dialplan (in default context)
exten => 3366,1,answer()
exten => 3366,n,playback(demo-instruct,noanswer)
exten => 3366,n,playback(demo-instruct,noanswer)
exten => 3366,n,playback(vm-goodbye,noanswer)
exten => 3366,n,hangup()
SIPP command
./sipp -l 399 -d 99000 -m 399 -s 3366 -p 5061 -sn uac 127.0.0.1 -trace_err
I was able to do 260 concurrent calls with no issues. The 2 playbacks for
demo-instruct were to cover 99 seconds since the file is only 67 seconds
long. For the 300/1000 call scenario, you would need to duplicate the line
accordingly. The limiting factor for me was my rtp.conf. I set up a range
of 10001-10520 which stopped at 260 since each "call" allocates 4 rtp slots
(2 in use and 2 for transfer, etc).
That's not quite correct. RTP ports are not allocated for 'transfers'. 2
ports are used for each media stream that can be used on a channel.
Since each channel has an audio stream, that will consume 2 ports. If
video support is enabled for the channel (even if it is not in use),
then 2 more ports will be consumed.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: [email protected] | SIP: [email protected] | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
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