[asterisk-users] Optimizing Asterisk Environment

[Andrew Latham]

On Sat, Mar 23, 2013 at 3:21 PM, Nick Khamis <symack at gmail.com> wrote:
> Hello Gentlemen,
>
> Thank you so much for your responses. We have been working on a
> SIP/RTP Proxy+Asterisk in backed by MySQL for a few weeks. Everything
> is working nicely I am pleased to say. And will be making some
> donations for G729 licenses etc.. (it's the least we can do to support
> the cause).
>
> Speaking about transcoding cards. We are functioning 100% on SIP using
> u/alaw and eventually G729. Some typical observations being great
> performance when not using G729 :)...
> Is there any transcoding happening when using only G729 and no other
> codec? We tried "disallow=all" and "allow=g729" and judging by the CPU
> load "260%" there seems to be...
>
> I hope this is not a silly question, but if we force the DID reseller
> to send only G729 encoded media, our asterisk server only allows G729,
> and finally for termination most sip trunk providers have g729 in
> there list of supported codecs, would there still be transcoding
> happening on our * box? I hope this is not as silly question as I
> think....
>
> To answer your question, we also tried with only ulaw and alaw and we
> seem to be stuck on exactly 101 peak. Is there a "limit" setting
> hidden in one of the "*.conf" files?
>
> We let sipp run for almost 3 hours on our box, from another local
> computer using the following command:
>
> <extensions.conf>
>
> exten => 1002,1,Answer
> exten => 1002,n,Goto(demo,s,1)
> exten => 1002,n,Hangup
>
> ./sipp -sn uac -d 10000 -s 1002 test.example.com -l 200 -mp 5606:
>
>
> And we got the following results: http://pastebin.com/J0YCprCb
>
> At 9.4 cps 96963 calls were executed with 0 failed calls. Where is the
> concurrent call figure in this tool? Please forgive me still getting
> use to it :).
>
> In regards to hardware transcoding cards for SIP protocol. Please let
> us know of some digium solutions. Again, we would love to support the
> cause.
>
> Nick.
>
> On 3/23/13, Andrew Latham <lathama at gmail.com> wrote:
>> On Sat, Mar 23, 2013 at 12:06 PM, Joshua Colp <jcolp at digium.com> wrote:
>>> Nick Khamis wrote:
>>>>
>>>> Oh no secret. Some things I do is increase the ulimit size. I was
>>>> wondering if there was a way to increase allocated memory. I have been
>>>> reading about a -p option but when I start asterisk using "asterisk -p
>>>> -10" it does not accept it but "asterisk -p 10" works fine. Not sure
>>>> if that was the intended new value.
>>>>
>>>> Also, I  just want to mention I am not trying to break any records.
>>>> Just would like to get a ~200 concurrent call stable environment using
>>>> G729 out of our setup.
>>>
>>>
>>> Are you transcoding? If so then that is where most of your CPU is going,
>>> and
>>> the only option to make it go further is to use a hardware transcoding
>>> solution.
>>>
>>> --
>>> Joshua Colp
>>> Digium, Inc. | Senior Software Developer
>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>>> Check us out at:  www.digium.com  & www.asterisk.org
>>
>> +1 on hardware card.  There are various other tools, even a network
>> based encoding solution.  Offloading to hardware can show you how
>> stable/strong your system might already be.
>>
>> --
>> ~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~

Are you recording calls?  If so that is a transcode if you are using
WAV or other.

-- 
~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~