[asterisk-users] Optimizing Asterisk Environment

[Eric Wieling]

It isn't a donation, it is a licensing fee so you can legally transcode g729.

If Asterisk has to modify or generate an audio stream, then you need to transcode.

Examples of this are early audio ringback, conferencing, playing back any audio files which are not already in g729 format, I'm sure there are others. 



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nick Khamis
Sent: Saturday, March 23, 2013 3:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Optimizing Asterisk Environment

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 ~
>
> --
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