<div dir="ltr"><div>Here is some good data on per call real bandwidth consumption for different codecs:</div><div> </div><div>Codec BR NEB<br>G.711 64 Kbps 87.2 Kbps<br>G.729 8 Kbps 31.2 Kbps<br>G.723.1 6.4 Kbps 21.9 Kbps<br>
G.723.1 5.3 Kbps 20.8 Kbps<br>G.726 32 Kbps 55.2 Kbps<br>G.726 24 Kbps 47.2 Kbps<br>G.728 16 Kbps 31.5 Kbps<br>iLBC 15 Kbps 27.7 Kbps<br><br>BR = Bit rate<br>NEB = Nominal Ethernet Bandwidth (one direction)</div>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Oct 23, 2013 at 5:33 AM, Miha <span dir="ltr"><<a href="mailto:miha@softnet.si" target="_blank">miha@softnet.si</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi Steve,<br>
<br>
do you have any good preposal which switch has good performance for voip<br>
trafic?<br>
<br>
miha<br>
<br>
Dne 10/22/2013 3:32 PM, piše Steve Underwood:<br>
<div class="HOEnZb"><div class="h5">> Hi Grant,<br>
><br>
> Two possibilities spring to mind:<br>
><br>
> - If your audio is coming from a disk, can that disk keep up?<br>
> - How good is your ethernet switch?<br>
><br>
> Notice in the second point I said how good, not how expensive. Many<br>
> switches choke on a large number of small media packets, including some<br>
> expensive big name products.<br>
><br>
> Regards,<br>
> Steve<br>
><br>
> On 10/22/2013 06:01 PM, Grant Bagdasarian wrote:<br>
>> The network shouldn’t be an issue, since we have at least 1Gbps lines.<br>
>> The tests stay within the network.<br>
>><br>
>> I forgot to mention the calls are being distributed across two<br>
>> machines by a Kamailio instance.<br>
>><br>
>> So for a total of 800 concurrent calls generated by Sipp, each machine<br>
>> has 400 active calls.<br>
>><br>
>> CPU load reaches about 70% per machine.<br>
>><br>
>> At this point both FS machines are virtualized, since the performance<br>
>> gain wasn’t that much compared to physical.<br>
>><br>
>> The VM host shows it is using ~3/4 of its CPU resources.<br>
>><br>
>> Htop shows that the normal priority threads(green) and the kernel<br>
>> threads(red) are about the same length.<br>
>><br>
>> Also, FS is running on Ubuntu Server 12.04 x64.<br>
>><br>
>> *From:*<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a><br>
>> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] *On Behalf Of<br>
>> *Stanislav Sinyagin<br>
>> *Sent:* Tuesday, October 22, 2013 11:11 AM<br>
>> *To:* FreeSWITCH Users Help<br>
>> *Subject:* Re: [Freeswitch-users] Audio quality issues<br>
>><br>
>> 800 calls at 64kbps is 51Mbps.<br>
>> Could there be a network issue, like a 100Mbps line between the endpoints?<br>
>><br>
>> How heavy is your CPU load? "htop" command would be helpful in this.<br>
>><br>
>> ------------------------------------------------------------------------<br>
>><br>
>> *From:*Grant Bagdasarian <<a href="mailto:GB@cm.nl">GB@cm.nl</a> <mailto:<a href="mailto:GB@cm.nl">GB@cm.nl</a>>><br>
>> *To:* "FreeSWITCH Users Help (<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
>> <mailto:<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>>)"<br>
>> <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
>> <mailto:<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>>><br>
>> *Sent:* Tuesday, October 22, 2013 10:14 AM<br>
>> *Subject:* [Freeswitch-users] Audio quality issues<br>
>><br>
>> Hello,<br>
>><br>
>> I was wondering what the maximum concurrent calls for FS before audio<br>
>> quality becomes an issue? I assume the specs of the machine would also<br>
>> affect this.<br>
>><br>
>> We are currently running FS on a Six Core (12 Threads) Intel E5-2430<br>
>> CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality<br>
>> at these rates is still fair, but we do notice some quality issue’s.<br>
>><br>
>> Going above these numbers screws up the audio quality: choppy sound,<br>
>> audio drops etc. We aren’t doing any heavy media processing, just<br>
>> simply playing a file (G711-Alaw) which lasts about 2 minutes during<br>
>> the load test.<br>
>><br>
>> These numbers are for one way audio, where Sipp doesn’t echo the RTP<br>
>> back. These numbers get lower once Sipp echo’s the RTP.<br>
>><br>
>> I’ve tried FS on a physical box and also on a virtual box (ESXi 5.1),<br>
>> but the performance gain on physical vs virtual isn’t that much.<br>
>><br>
>> I disabled all the modules we don’t need, like CDR’s, conferencing, etc.<br>
>><br>
>> Are there any parameters(config files)/modules that can affect the<br>
>> quality of the audio stream?<br>
>><br>
>> Regards,<br>
>><br>
>> Grant<br>
>><br>
>><br>
><br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Guillermo Ruiz Camauer<br>
</div>