[Freeswitch-users] Audio quality issues

[Anthony Minessale]

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 at 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 at yahoo.com>
> Date: Oct 22, 2013 4:12 PM
> Subject: Re: [Freeswitch-users] Audio quality issues
> To: "FreeSWITCH Users Help" <freeswitch-users at 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 at cm.nl>
> *To:* "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" <
> freeswitch-users at 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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>
> 
> 
>
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>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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>
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>


-- 
Anthony Minessale II

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