That was a memorable milestone in FS development history the first time we broke that barrier and got confused why it was not working.... =D Before that it was unheard of to saturate 100mbps with existing voip software..... On Tue, Oct 22, 2013 at 5:58 PM, Donny Hardyanto <hardyanto.donny@gmail.com>wrote:
Actually G711 will consumes abaout 167kbps for a call (64 kbps for each side, and the other is IP header, RTP header, Ethernet header) so 800 calls of this will be 133.8 Mbps. If your line is 100 Mbps then you have saturated the line.
Donny
---------- Forwarded message ---------- From: "Stanislav Sinyagin" <ssinyagin@yahoo.com> Date: Oct 22, 2013 4:12 PM Subject: Re: [Freeswitch-users] Audio quality issues To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Cc:
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.
------------------------------ *From:* Grant Bagdasarian <GB@cm.nl> *To:* "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org)" < freeswitch-users@lists.freeswitch.org> *Sent:* Tuesday, October 22, 2013 10:14 AM *Subject:* [Freeswitch-users] Audio quality issues
Hello,
I was wondering what the maximum concurrent calls for FS before audio quality becomes an issue? I assume the specs of the machine would also affect this. We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality at these rates is still fair, but we do notice some quality issue’s. Going above these numbers screws up the audio quality: choppy sound, audio drops etc. We aren’t doing any heavy media processing, just simply playing a file (G711-Alaw) which lasts about 2 minutes during the load test. These numbers are for one way audio, where Sipp doesn’t echo the RTP back. These numbers get lower once Sipp echo’s the RTP.
I’ve tried FS on a physical box and also on a virtual box (ESXi 5.1), but the performance gain on physical vs virtual isn’t that much.
I disabled all the modules we don’t need, like CDR’s, conferencing, etc.
Are there any parameters(config files)/modules that can affect the quality of the audio stream?
Regards,
Grant
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