[asterisk-users] Asterisk V/s FreeSwitch

[Kevin P. Fleming]

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
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