[asterisk-users] high cpu average load
Kamlesh Kumar
kamlesh_kmr at hotmail.com
Fri Sep 6 05:33:54 CDT 2013
Date: Thu, 5 Sep 2013 12:11:36 -0700
From: asterisk.org at sedwards.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] high cpu average load
On Thu, 5 Sep 2013, Kamlesh Kumar wrote:
> Running one asterisk server with below details.
> Only SIP to SIP calls. No real time configuration, no recording, no voicemail, no IVR, no codec translation. Average CPU load varies between 4 to 30 for 150 to 200 concurrent calls and
> we start getting problem in call quality like delay in connectivity, voice breakage etc....
>
> Hardware:
> 2 Physical processor Intel(R) Xeon(R) CPU 5120 @ 1.86GHz
> 8 GB RAM
> 500 GB Sata HDD
>
> Asterisk: 1.6.2.9
> PHP 5.3.3 (cli)
> MySQL: 5.0.77
> Linux: CnetOS 5.5 (Final)
>
> Please suggest the solution.
Need a bit more detail.
The 5120 is kind of a wimpy processor, but what is keeping it busy?
What do 'top' and 'htop' show are consuming the processor?
What is your application?
What are 200 calls doing?
Are you calling a bunch of AGIs written in scripting languages?
Eliminating translation is difficult. How do you know you were successful?
Do 'module show like codec_' and 'module show like format_' show anything
unexpected?
Below are the further details:top and htop shows that 'asterisk' is consuming the whole cpu power.Application: Kind of SIP trunking - call is coming from IP and using dialplan routed to other third party IPAre you calling a bunch of AGIs written in scripting languages?
only one AGI script written in PHP is called with 'h' extension once the call is hungupmodule show like codec_
vm*CLI> module show like codec_
Module Description Use Count
codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0
codec_dahdi.so Generic DAHDI Transcoder Codec Translato 0
codec_alaw.so A-law Coder/Decoder 0
codec_lpc10.so LPC10 2.4kbps Coder/Decoder 0
codec_adpcm.so Adaptive Differential PCM Coder/Decoder 0
codec_g722.so ITU G.722-64kbps G722 Transcoder 0
codec_g726.so ITU G.726-32kbps G726 Transcoder 0
codec_ulaw.so mu-Law Coder/Decoder 0
codec_gsm.so GSM Coder/Decoder 0
9 modules loadedmodule show like format_
vm*CLI> module show like format_
Module Description Use Count
format_g726.so Raw G.726 (16/24/32/40kbps) data 0
format_ogg_vorbis.so OGG/Vorbis audio 0
format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0
format_g729.so Raw G729 data 0
format_wav.so Microsoft WAV format (8000Hz Signed Line 0
format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0
format_g723.so G.723.1 Simple Timestamp File Format 0
format_gsm.so Raw GSM data 0
format_vox.so Dialogic VOX (ADPCM) File Format 0
format_mp3.so MP3 format [Any rate but 8000hz mono is 0
10 modules loaded
Thank you,Kamlesh
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130906/328515e9/attachment.htm>
More information about the asterisk-users
mailing list