Hello, please keep the list in CC.
Ok, the machine seems to not really using all its cores. Regarding playing a file, this can be also done with rtpengine: https://kamailio.org/docs/modules/5.5.x/modules/rtpengine.html#rtpengine.f.play_media Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Mohammad Reza Keshavarzianpoor <[email protected]> Sent: Monday, January 16, 2023 3:35 PM To: Henning Westerholt <[email protected]> Subject: Re: [SR-Users] rtp lost package in 300 concurrent calls and above thanks for your reply my kamailio server has 8GB of ram and 4core*2.6 on esxi(dedicated)Ubuntu 22.04.1 LTS. is it low? htop on server shows load average 1.04 at max and I didn't see more than 10% load on any core of the server. I've selected rtp media server because of playback of a voice file over incoming calls in the future usage On Mon, Jan 16, 2023 at 5:45 PM Henning Westerholt <[email protected]<mailto:[email protected]>> wrote: Hello, do you observe (too) high load on the system with the standard system management tools? If you want to just proxy calls, my recommendation would be to use rtpengine. Here you should be able to achieve higher concurrent calls rates, if you have a few CPUs on a decent machine. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Mohammad Reza Keshavarzianpoor <[email protected]<mailto:[email protected]>> Sent: Monday, January 16, 2023 2:50 PM To: [email protected]<mailto:[email protected]> Subject: [SR-Users] rtp lost package in 300 concurrent calls and above Hi, I've installed last version of kamailio over ubuntu. I need to broadcast machine calls to destinations. my concurrent call should be around 300(my pstn channels), so I start calls with cps around 20, because my calls contains a less than 30 second and if destination answer the call it will hanged up in less than or equal to 30 seconds. My problem is: to test my kamailio config, I'm using startrinity sip tester over an win server 2019 on local network. and it shows about 20% lost package on rtp and a jitter about 500ms. what should I do? my config is based on rtp media server module. Is there a complete working sample config for kamailio with rtp media server module? I tried rtpproxy and rtp engine in past and the same problem happened. Thanks for any help you can provide Best regards
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