On Sat, Mar 23, 2013 at 3:21 PM, Nick Khamis <[email protected]> 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 <[email protected]> wrote: >> On Sat, Mar 23, 2013 at 12:06 PM, Joshua Colp <[email protected]> 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 [email protected] http://lathama.net ~
Are you recording calls? If so that is a transcode if you are using WAV or other. -- ~ Andrew "lathama" Latham [email protected] http://lathama.net ~ -- _____________________________________________________________________ -- 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
