[asterisk-users] Asterisk V/s FreeSwitch

[virendra bhati]

Thanks for reply and share your techniques, dialplans and knowledge on this
thread. But my question was not related to load-balancing. I want to know ,
Why freeSwitch can preferred with compare to Asterisk(Call base , quality
base)? And what is architecture difference between them.


I am totally agree that by using SIPp we can not relay that server can
handle so much load. because by using MOH only CPU load can major and we
can check how many thread asterisk can open.

On Fri, Feb 10, 2012 at 2:34 AM, Kevin P. Fleming <kpfleming at digium.com>wrote:

> 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: kfleming at digium.com | SIP: kpfleming at digium.com | 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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-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbhati at gmail.com
Skype id:- virbhati2
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