<div dir="ltr">We are currently at around 100 concurrent calls. The load is fairly stable during use and not very "peaky". We would like to be able to double the amount to about 200 simultaneous calls.<div><br></div>
<div>As for recording we would, if performance allows, initially want to let freeswitch record directly to our selected format (i.e. mp3 or ogg). And if we later get issues make that process more efficient (lazy transcoding). </div>
<div><br></div><div>The WebRTC would be only for internal use where we have complete control of the environment. It would be used to let users authenticate and connect their browsers to FSW. WebRTC would also be a second step and not something we would implement immediately.</div>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Jun 5, 2014 at 12:06 AM, Kristian Kielhofner <span dir="ltr"><<a href="mailto:kris@kriskinc.com" target="_blank">kris@kriskinc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">With your setup I see two things about WebRTC to consider from a<br>
performance perspective:<br>
<br>
1) Codec<br>
2) SRTP<br>
<br>
WebRTC endpoints (more than likely) will be using Opus. Even though<br>
they support PCMU and PCMA the use of these codecs is ill-advised<br>
considering the likely state of the endpoint and network. WebRTC also<br>
mandates the use of SRTP.<br>
<br>
Both of these have a potentially significant impact on your system<br>
because you will be recording. This means that your system will have<br>
to decrypt the SRTP stream and transcode it to store to common<br>
recording formats (such as WAV). Even still, with the kind of volume<br>
and applications you're describing Mike's comments earlier were spot<br>
on.<br>
<br>
For other applications and uses there are all kinds of additional<br>
things to consider (like the impact of ICE negotiation on CPS, for<br>
example).<br>
<div><div class="h5"><br>
On Wednesday, June 4, 2014, Oscar Franzen <<a href="mailto:oscar.franzen@advisa.se">oscar.franzen@advisa.se</a>> wrote:<br>
><br>
> Thanks!<br>
><br>
> I forgot to mention that we want to use WebRTC in the near future as well. Is there anything special I have to consider for that?<br>
><br>
> So you basically think the IO for recordings is going to be the main bottleneck?<br>
> I am not entirely sure I can do transcoding at the storage server but I'll look into it.<br>
><br>
> As I understand it FSW has to do some sort of transcoding when mixing two channels to record them? Is it possible / would you recommend to record to g711 (I think this is the used codec)? And then later transcode it to mp3 for long time storage?<br>
><br>
> What other bottlenecks am I looking at? What are some usual performance problems and when do you usually encounter them?<br>
> My initial idea was to go with a modern multi core Intel Xeon processor and multiple network interfaces. Is the next thing to focus on storage performance?<br>
><br>
> Oscar<br>
><br>
><br>
> On Wed, Jun 4, 2014 at 9:27 PM, Guillermo Ruiz Camauer <<a href="mailto:grcamauer@gmail.com">grcamauer@gmail.com</a>> wrote:<br>
><br>
> I would concentrate on the disk side of things with a good stripe or RAID of SSDs. Or, just add enough RAM (say 64GB) and make a large enough tmpfs/RAMdisk partition to store your recordings before transcoding and shipping them off to the storage server. Just make sure to have UPS backed power.<br>
> Can the transcoding be done on the storage server (they usually have lots of spare CPU cycles)? Separate network adapters for VoIP and storage would also be good.<br>
> I would go with a CPU with 4 cores.<br>
><br>
> Testing will always be the final answer.<br>
><br>
> Guillermo<br>
><br>
><br>
> On Wed, Jun 4, 2014 at 3:30 PM, Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>> wrote:<br>
><br>
> Not a whole lot of cpu from any modern non embedded processor can probably handle this. Test to see exactly how it behaves with your exact requirements. Sound card would do nothing at all for transcoding, thats all cpu.<br>
><br>
> Mike<br>
><br>
> On Jun 4, 2014, at 6:20 PM, Oscar Franzen <<a href="mailto:oscar.franzen@advisa.se">oscar.franzen@advisa.se</a>> wrote:<br>
><br>
> Hi!<br>
><br>
> I'm looking for some help with hardware for a production setup of freeswitch. I need at least 150 concurrent calls, but would prefer to buy a server which can handle more to facilitate growth.<br>
><br>
> I am looking at using the same codec (i think it is g711) for both legs of the call. I want to record all calls. Either to whatever codec and then transcode them separately or directly to mp3/ogg. I am going to use ESL to control FSW and also listen to most events over the socket.<br>
><br>
> What would you recommend for hardware? Is it recommended to buy a sound card for transcoding or just buy enough CPU? I will probably run from SSD, record to same SSD or other local disk and continuously move recordings to external (network) storage.<br>
><br>
> So:<br>
> 150 concurrent calls<br>
> ~10 initiations/second<br>
> mod_event_socket - for control and events<br>
> A limited amount of eavesdropping<br>
> recording (either to original codec or ogg/mp3)<br>
> Some "standard" modules.<br>
><br>
> Regards<br>
> Oscar<br>
><br>
><br>
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</div></div>> Regards<br>
> Oscar Franzén<br>
> Developer<br>
<div class="im HOEnZb">><br>
><br>
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</div></div></blockquote></div><br><br clear="all"><div><br></div>-- <br><div dir="ltr"><div><font color="#000000" face="arial, helvetica, sans-serif">Regards</font></div><font face="arial, helvetica, sans-serif" color="#000000"><b>Oscar Franzén<br>
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Mobil: 070-720 45 81<br>Besöksadress: </font>Kungsgatan 30, Stockholm</div><div><font color="#000000"><font face="arial, helvetica, sans-serif"><a href="http://www.advisa.se/" style="color:rgb(17,85,204)" target="_blank">www.advisa.se</a></font></font></div>
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