Need your opinion!

[Speedy2k]

I am building a new Asterisk box, and i would like to know how many active call i will be able to make before voice degradation?

Here is the PBX i'm building i want to put a AMD 2.4 X2 precessor with 1gig of ram and a 1000mbits network card.

My question is what i can expect with a 1000mbits link and with a 100mbits link. And does i really need a hard drive for that or just a IDE to SD adapter with a 2gig fast SD card would do the trick?? i would like to remove all the unecesary moving part and that make it easier for restauration thanx for your insight!!

 

[cjkeeme]

Your bottleneck isn't your network card, but your available bandwidth.

To be safe I would say you need about 100kb/s for each concurrent call. Also, take into consideration normal bandwidth usage from VPN, web browsing, email, etc.

Hardware wise you are fine, but I whole-heartedly recommend the Intel Atom as a base platform. I have an office of 38 extensions running on this architecture and it is flawless. It's even the single-core version, although the dual-core will be in my mailbox today.

As far as the IDE to SD converter you will need to speak with someone else on that. I always use hard disks.

 

[GregKeys]

Keep in mind, there are many factors that affect system performance.
What Codec(s) are you using? Some are easy on the processor, some are hard.
Are you using POTS, PRI, or SIP trunks? If SIP, what is your internet bandwidth (up and down)?
Will you be using the same codec throughout, or a mixed bag, which causes processor load due to translating from one codec to another?
What kind of switch are you using?
How much traffic is already on your network?
Will you be recording calls? All or on demand?
Conference calls with meetme?

My point is, that there are MANY factors that play a part in the "formula" for number of concurrent calls.

Greg Keys

 

[Speedy2k]

OK thanx a lot for those answer, i really appreciate. i'm looking to be able to run between 30 to 50 phone on this network with a mix of POTS and SIP lines. Which codec do you recommand?

 

[Linetux]

Bandwidth ...

Bandwidth is a funny thing... and it really does depend on a lot of factors.

We have an office full of around 80 phones. I've been monitoring our bandwidth use on the trunk ports back to the core switch - there are 2x48 port PoE switches involved in serving up all the phones. Each trunk will peak out around 300K/sec - so figure the big peaks aren't captured quite right and we're talking 500K x2, or 1Mbit/sec of traffic for 80 phones.

Don't mess with codecs. You have plenty of bandwidth, and they'll just put a load on the CPU you don't need to do.

Your biggest worry should be on the latency on your internet pipe to your SIP provider, not worrying about internal stuff. You could run 50 phones off of a VIA C7 CPU and a 10M connection without issue.

 

[jroper]

Your biggest worry should be on the latency on your internet pipe to your SIP provider

Swissscom surveys showed that the latency was not an issue untlil lag hit 350+ms - that's lag, not ping times, suggesting that the human ear, can put up with about 700mns ping times to the sip provider.

Joe

 

[GregKeys]

Speedy,
You reference the number of extensions, but this is not the critical factor.
1. Number of concurrent calls passing thru the server. This is affected by all the things mentioned in my previous email. You can have thousands of extensions with no real impact, but active simultaneous calls is a whole different story.

2. How many active SIP calls via trunks vs the internet bandwidth available. Local bandwidth is not so much and issue as a 100mbit connection can do somewhere over 1000 ulaw conversations. The math works out to 90k per call = 17 per T1 (1.5mbit) or 66 times that for 100mbit = 1133
In the real world, you do not get perfect use of available bandwith by a long shot but you are still WAY up there on the local network connection.
The internet connection on the other hand can really be limited at 90k per call. If you use one of the other codecs, you start trans-coding between ulaw for the internal calls and some other codec for the SIP Trunk calls and that starts loading your CPU.
Welcome to the circus It is the high wire balancing act of the IT world!

Greg Keys